From tleyden at branchcut.com  Mon Dec  1 00:31:28 2008
From: tleyden at branchcut.com (Traun Leyden)
Date: Mon, 1 Dec 2008 13:01:28 +0430
Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 29, Issue 189
In-Reply-To: <mailman.1445.1228105114.18013.freeswitch-users@lists.freeswitch.org>
References: <mailman.1445.1228105114.18013.freeswitch-users@lists.freeswitch.org>
Message-ID: <f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>

>
> Message: 9
> Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST)
> From: Marc Orenberg <marc at kasteris.com>
> Subject: [Freeswitch-users] Problem importing modules in mod_python
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <195670.44941.qm at web50805.mail.re2.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> In the latter versions of mod_python, I'm unable to import standard python
> modules such as time and MySQLdb.?
> For example, the following script works fine in version 1.0.1:
>
> ??? ??? import time
> ??? ??? import os
> ??? ??? from freeswitch import *
> ??? ??? def handler(session, args):
> ??? ??? ??? session.answer()
> ??? ??? ??? session.execute("sleep", "2000")
> ??? ??? ???
> session.streamFile("/usr/local/freeswitch/prompts/01Welcome.wav")
> ??? ??? ??? return(session)
>
> But in freeswitch-1.0.latest.tar.gz, and svn versions 10556-10558, I get
> the following error:
>
> ??? ??? 2008-11-30 21:13:09 [ERR] mod_python.c:129 eval_some_python() Error
> reloading module
> ??? ??? Traceback (most recent call last):
> ??????? File "/usr/lib/python2.4/site-packages/scripts/test.py", line 1, in
> ?
> ??? ??? import time
> ??? ??? ImportError: /usr/lib/python2.4/lib-dynload/timemodule.so:
> undefined symbol: PyExc_ValueError
>
> Thanks for your help!
>

I have run into the same problem and put some documentation on the wiki:

http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2F...2Fdatetime.so:_undefined_symbol:_PyExc_IOError

I think something changed in freeswitch in the way it is loading modules, or
at least the way it is loading mod_python.  This behavior appeared all of
the sudden in recent freeswitch versions.

HTH
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From yudha2008 at gmail.com  Mon Dec  1 02:54:25 2008
From: yudha2008 at gmail.com (Baskar)
Date: Mon, 1 Dec 2008 16:24:25 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
Message-ID: <d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>

Hi,
 *
   **It is possible to dial outbound through console dialing. Yes means me
How ?**

   Without using the softphone how can i dial outbound from freeswitch
console itself. *
*
   I want to Know without using any softphone for calling.

   It is possible in asterisk. we can dial from console itself. *

*   So i want to know it is possible in freeswitch.*

Warm Regards,
N.Baskar
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From gmaruzz at celliax.org  Mon Dec  1 03:15:34 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 1 Dec 2008 12:15:34 +0100
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
Message-ID: <7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>

Hello Baskar,

in FS it is possible to call from console using the endpoint mod_portaudio.

Please have a look at
http://wiki.freeswitch.org/wiki/Freeswitch_softphone , it is *NOT REAL
SOFTPHONE* it is FS used *LIKE* a softphone. Exactly as in Asterisk
with chan_alsa or chan_oss.

Sincerely,

Giovanni Maruzzelli

=========================================
Contact person : Mr Giovanni Maruzzelli
Company : celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Phone : 39-347-2665618
Fax : 39-02-87390039



On Mon, Dec 1, 2008 at 11:54 AM, Baskar <yudha2008 at gmail.com> wrote:
> Hi,
>
>    It is possible to dial outbound through console dialing. Yes means me How
> ?
>
>    Without using the softphone how can i dial outbound from freeswitch
> console itself.
>
>    I want to Know without using any softphone for calling.
>
>    It is possible in asterisk. we can dial from console itself.
>
>    So i want to know it is possible in freeswitch.
>
> Warm Regards,
> N.Baskar
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From Prometheus001 at gmx.net  Mon Dec  1 03:39:06 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 01 Dec 2008 12:39:06 +0100
Subject: [Freeswitch-users] Inbound 1-way audio issue using GSM codec
In-Reply-To: <000301c951ba$9705dac0$c5119040$@com>
References: <49305CBF.8060801@ieee.org> <000301c951ba$9705dac0$c5119040$@com>
Message-ID: <4933CCDA.40905@gmx.net>

Hello Maxim,

can you reach another internal device except the GSM one in order to see
whether it's GSM codec specific?

However I can see that you're using local IPs (10.x.x.x) so I expect
that they are natted. This often causes one way audio when the external
rtp-ip is not set. Please try to set a
    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
entry to internal.xml and external.xml in your SIP profiles and see if
it works. Use stun at least for the internal profile (FQDN and external
IP most probably will not work)

Best regards
Peter

Maxim Karp schrieb:
> Hello,
>
> I am using a GSM based endpoint connected to freeswitch that makes calls to
> the PSTN via a SIP gateway (SBC).  The SBC uses PCMU between itself and
> freeswitch.
>
> When I make an outgoing call from a GSM based device via freewsitch to the
> PSTN via the SBC, everything works fine and audio works in both directions
> for both end points.  I looked at the console logs and they do indicate that
> I am using GSM.
>
> Console output when I dial and before answer on the GSM device:
>
> v=0
> o=- 74 0 IN IP4 10.229.0.58
> s=session
> c=IN IP4 10.229.0.58
> b=CT:17
> t=0 0
> m=audio 59806 RTP/AVP 8 0 3 97 101
> k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 RED/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=encryption:optional
>
> Console output once it rings and after I answer on the PSTN side:
>
> v=0
> o=FreeSWITCH 1227887572 1227887573 IN IP4 10.229.0.10
> s=FreeSWITCH
> c=IN IP4 10.229.0.10
> t=0 0
> a=sendrecv
> m=audio 30896 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> When I receive a call from the SIP gateway, the endpoint making the call
> (not on freeswitch) can't hear me speaking from the GSM device connected to
> freeswitch.  I can hear everything fine on the GSM device.
>
> Here is the console output for the call info coming in from the PSTN.
>
> v=0
> o=FreeSWITCH 1227902084 1227902085 IN IP4 38.113.164.132
> s=FreeSWITCH
> c=IN IP4 38.113.164.132
> t=0 0
> a=sendrecv
> m=audio 16724 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> Here is how I have vars.xml configured:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
>
> When I prioritize GSM on the outbound codec prefs I get static on the PSTN
> side. 
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/>
>
> Any ideas?
>
> Maxim.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From odermann at googlemail.com  Mon Dec  1 03:46:39 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 1 Dec 2008 12:46:39 +0100
Subject: [Freeswitch-users] Leg A terminated by Leg B on a uuid_bridge
In-Reply-To: <872970CF4A55BF42A5337D570860209F01052E34@HPEXCHVS01.exchange.airg>
References: <872970CF4A55BF42A5337D570860209F01052E34@HPEXCHVS01.exchange.airg>
Message-ID: <5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>

hi  simon,

i am not sure, if i understood your problem right, but if you do not
want leg a to hang up after leg b (the originated call) hangs up, set
"park_after_bridge=true" when you make the originate.
as far as i know, "hangup_after_bridge=false" is only for the inbound
and helps nothing with the outbound.

if you want something different, please explain me a little more.

dennis



2008/11/28 Simon Tang <simon at airg.com>:
> Hello,
>
>
>
> I'm using event socket outbound, and have an issue where, after a bridge
> ends and is terminated by Leg B, Leg A is also terminated.  Here's the call
> flow:
>
>
>
> 1.       Call comes in (Leg A), session created, play welcome message.
>
> 2.       From this session, originate and dial out using api originate
>
> 3.       After the target answers (Leg B), bridge the 2 calls using api
> uuid_bridge
>
> 4.       Leg B hangs up.
>
> 5.       Leg A will be terminated.
>
>
>
> After step 4, Leg A is terminated.  I do not want Leg A to hang up.  I've
> tried setting "hangup_after_bridge=false" prior to the call, and that
> doesn't work.
>
>
>
> Having said that, I tried a similar test which does not end Leg A's call
> after Leg B hangs up, but I can't use this solution because, functionally,
> does not accomplish what I want it to do (i.e., I want to perform some
> actions on Leg B prior to the bridge, like send some DTMF tones, playback
> some messages, etc).  I did not need to set the "hangup_after_bridge"
> variable (default should be false anyway).
>
>
>
> 1.       Call comes in (Leg A), session created, play welcome message.
>
> 2.       From this session, do a bridge by doing an execute bridge.
>
> 3.       The target answers (Leg B)
>
> 4.       Leg B hangs up.
>
> 5.       Leg A will still be active.
>
>
>
> Any ideas would be appreciated.  Thanks!
>
>
>
> Simon
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From saigop at gmail.com  Mon Dec  1 04:25:31 2008
From: saigop at gmail.com (Gopala krishnan)
Date: Mon, 1 Dec 2008 17:55:31 +0530
Subject: [Freeswitch-users] Leg A terminated by Leg B on a uuid_bridge
In-Reply-To: <5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>
References: <872970CF4A55BF42A5337D570860209F01052E34@HPEXCHVS01.exchange.airg>
	<5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>
Message-ID: <2ea4d47e0812010425i1f278768i2be711c74a2e00b8@mail.gmail.com>

Hi Simon,
  You can get the A leg uuid and B leg uuid seperately and can hangup
whichever the leg you need...:)



-- 
Thank you  with regards,
Gopal,
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From ttroy50 at gmail.com  Mon Dec  1 05:09:57 2008
From: ttroy50 at gmail.com (matrim)
Date: Mon, 1 Dec 2008 05:09:57 -0800 (PST)
Subject: [Freeswitch-users]  TLS receiving calls
Message-ID: <20771637.post@talk.nabble.com>


Hi,

I'm having problems using TLS to receive calls.

I'm using a Nokia N95 to test TLS against freeswitch. I can register my
client against freeswitch and make outbound calls to the test numbers (e.g.
9999).

I can also make calls to other users registered over UDP.

However if I try to make a call to a user registered over TLS the leg of the
call to that user always goes via UDP.

e.g.

1000 registered via TLS
1001 registered via TLS
1002 registered via UDP
1003 registered via UDP

1000 -> 1002 works ok
1003 -> 1002 works ok

1001 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000
tries to setup via UDP
1002 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000
tries to setup via UDP

===

>From looking at some of the documentation it seems to me that the issue may
be with the "tls-bind-params" being "transport=tls".

The phone I'm using doesn't add the "transport=tls" parameter, and only uses
"sips:" to specify that the connection is via TLS.

I tried setting "tls-bind-params" to a blank string but it didn't change
anything. Is there any way to receive calls over TLS if you don't specify
"transport=tls" in your contact string during registration?

According to RFC3261 the use of the "transport=tls" parameter isn't
recommended anymore and is now deprecated.


-- 
View this message in context: http://www.nabble.com/TLS-receiving-calls-tp20771637p20771637.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From fs_ask_sy at citromail.hu  Mon Dec  1 05:35:39 2008
From: fs_ask_sy at citromail.hu (x y)
Date: Mon, 01 Dec 2008 14:35:39 +0100
Subject: [Freeswitch-users] Multi FS behind same NAT,
	PRACK goes to wrong port
In-Reply-To: <5FD8C155-AFD9-451E-B58D-31CC47CB2EA6@freeswitch.org>
Message-ID: <20081201133539.27255.qmail@server15.citromail.hu>

            Hy!

You were right about the contact in 183, its port 5060 in there.
I've tried turning of 100rel, it seemed to work with calls, but caused some problems with others things, so I would really appreciate if there is another option.
Btw, I have mentioned that, that I had gateway problems too. Setting up ext-ip as stun.freeswitch.org has seemed to work, but after 5 days, the gateway has went down again with the same 503 error. Is there any common in the two issues?

Thx for your advices.

Cheers,
Viktor





################################################################
U xxx.xxx.xxx.xxx:56965 -&gt; yyy.yyy.yyy.yyy:5060
INVITE sip:252252%233619995384 at box.net:5060 SIP/2.0.
Via: SIP/2.0/UDP  xxx .xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" &lt;sip:xxx.xxx.xxx.xxx&gt;;tag=7185D258-BB0.
To: &lt;sip:252252%233619995384 at box.net&gt;.
Date: Thu, 27 Nov 2008 16:28:41 GMT.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313 at xxx.xxx.xxx.xxx.
Supported: timer,100rel.
Min-SE:  1800.
Cisco-Guid: 1234720477-3151434205-2374282417-4237312787.
User: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO.
CSeq: 101 INVITE.
Max-Forwards: 10.
Remote-Party-ID: &lt;sip:xxx.xxx.xxx.xxx&gt;;party=calling;screen=yes;privacy=full.
Timestamp: 1227803321.
Contact: &lt;sip:xxx.xxx.xxx.xxx:5060&gt;.
Expires: 180.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 264.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 5202 8450 IN IP4 xxx.xxx.xxx.xxx.
s=SIP Call.
c=IN IP4 xxx.xxx.xxx.xxx.
t=0 0.
m=audio 16732 RTP/AVP 3 8 101.
c=IN IP4 xxx.xxx.xxx.xxx.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.

#
U yyy.yyy.yyy.yyy:5060 -&gt; xxx.xxx.xxx.xxx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" &lt;sip:xxx.xxx.xxx.xxx&gt;;tag=7185D258-BB0.
To: &lt;sip:252252%233619995384 at box.net&gt;.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313 at xxx.xxx.xxx.xxx.
CSeq: 101 INVITE.
Timestamp: 1227803321 0.000388.
User-Agent: agent
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -&gt; zzz.zzz.zzz.zzz:1352
INVITE sip:3619995384 at zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F.
Max-Forwards: 8.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
Contact: &lt;sip:usr at yyy.yyy.yyy.yyy:5060;transport=udp&gt;.
User-Agent: agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 398.
Remote-Party-ID: "00000000" &lt;sip:00000000 at dom&gt;;screen=yes;privacy=full.
.
v=0.
o=FreeSWITCH 6476130113585053783 8141266268953030291 IN IP4 yyy.yyy.yyy.yyy.
s=FreeSWITCH.
c=IN IP4 yyy.yyy.yyy.yyy.
t=0 0.
a=sendrecv.
m=audio 17068 RTP/AVP 3 98 8 9 0 18 101 13.
a=rtpmap:3 GSM/8000.
a=rtpmap:98 SPEEX/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U zzz.zzz.zzz.zzz:1352 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
User-Agent: agent2
Content-Length: 0.
.

###
U zzz.zzz.zzz.zzz:1352 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
Contact: &lt;sip:mod_sofia at zzz.zzz.zzz.zzz:5060;transport=udp&gt;.
RSeq: 2093511444.
User-Agent: agent2
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Require: 100rel.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 264.
.
v=0.
o=FreeSWITCH 6247558966294607749 119302723364474833 IN IP4 zzz.zzz.zzz.zzz.
s=FreeSWITCH.
c=IN IP4 zzz.zzz.zzz.zzz.
t=0 0.
m=audio 24756 RTP/AVP 3 101 13.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U yyy.yyy.yyy.yyy:5060 -&gt; zzz.zzz.zzz.zzz:5060
PRACK sip:mod_sofia at zzz.zzz.zzz.zzz:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK5DH0ryBH70FSB.
Max-Forwards: 70.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783325 PRACK.
Contact: &lt;sip:usr at yyy.yyy.yyy.yyy:5060;transport=udp&gt;.
RAck: 2093511444 107783324 INVITE .
User-Agent: agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Content-Length: 0.
.

#
U zzz.zzz.zzz.zzz:5060 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 481 No such response.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5DH0ryBH70FSB.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783325 PRACK.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -&gt; zzz.zzz.zzz.zzz:1352
CANCEL sip:3619995384 at zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F .
Max-Forwards: 8.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 CANCEL.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -&gt; xxx.xxx.xxx.xxx:5060
SIP/2.0 481 Call/Transaction Does Not Exist.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" &lt;sip:xxx.xxx .xxx.xxx&gt;;tag=7185D258-BB0.
To: &lt;sip:252252%233619995384 at box.net&gt;;tag=cXQ5m2QSmU2QB.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313 at xxx .xxx.xxx.xxx.
CSeq: 101 INVITE.
User-Agent: agent
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary.
Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE".
Content-Length: 0.
.

#
U xxx.xxx.xxx.xxx:56965 -&gt; yyy.yyy.yyy.yyy:5060
ACK sip:252252%233619995384 at box.net:5060 SIP/2.0.
Via: SIP/2.0/UDP  xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" &lt;sip:xxx.xxx.xxx.xxx&gt;;tag=7185D258-BB0.
To: &lt;sip:252252%233619995384 at box.net&gt;;tag=cXQ5m2QSmU2QB.
Date: Thu, 27 Nov 2008 16:28:41 GMT.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313 at xxx.xxx.xxx.xxx.
Max-Forwards: 10.
Content-Length: 0.
CSeq: 101 ACK.
.

#
U zzz.zzz.zzz.zzz:1352 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 CANCEL.
Content-Length: 0.
.

#
U zzz.zzz.zzz.zzz:1352 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
User-Agent: agent2
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -&gt; zzz.zzz.zzz.zzz:1352
ACK sip:3619995384 at zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F.
Max-Forwards: 8.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 ACK.
Content-Length: 0.
.

#########################################################





Hirdet?s (x)      


RENDELJ MOST! - H?ztart?si g?peket AKCI?S ?ron!


T?bb mint 300 VIDE? term?kbemutat? seg?t v?lasztani, ak?r 5 ?V kiterjesztett garanci?val rendelhetsz ITT!  

S?t?-f?z?lap szettek, mos?g?pek, mosogat?g?pek, t?zhelyek - ORSZ?GOS sz?ll?t?ssal a
MARKABOLT.hu-t?l.
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From regs at kinetix.gr  Mon Dec  1 06:01:40 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Mon, 01 Dec 2008 16:01:40 +0200
Subject: [Freeswitch-users] Set variable for the outgoing leg
Message-ID: <4933EE44.60900@kinetix.gr>

All the variables that I set show up only in the a-leg CDR.
How can I set a variable that can be used during the b-leg CDR generation?


From Prometheus001 at gmx.net  Mon Dec  1 07:47:00 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 01 Dec 2008 16:47:00 +0100
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <20771637.post@talk.nabble.com>
References: <20771637.post@talk.nabble.com>
Message-ID: <493406F4.204@gmx.net>

Did you add
<action application="export" data="sip_secure_media=true"/>
into youy dialplan before bridging that call. How is your internal.conf,
is TLS enabled there?

Best regards
Peter

matrim schrieb:
> Hi,
>
> I'm having problems using TLS to receive calls.
>
> I'm using a Nokia N95 to test TLS against freeswitch. I can register my
> client against freeswitch and make outbound calls to the test numbers (e.g.
> 9999).
>
> I can also make calls to other users registered over UDP.
>
> However if I try to make a call to a user registered over TLS the leg of the
> call to that user always goes via UDP.
>
> e.g.
>
> 1000 registered via TLS
> 1001 registered via TLS
> 1002 registered via UDP
> 1003 registered via UDP
>
> 1000 -> 1002 works ok
> 1003 -> 1002 works ok
>
> 1001 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000
> tries to setup via UDP
> 1002 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000
> tries to setup via UDP
>
> ===
>
>   
>> >From looking at some of the documentation it seems to me that the issue may
>>     
> be with the "tls-bind-params" being "transport=tls".
>
> The phone I'm using doesn't add the "transport=tls" parameter, and only uses
> "sips:" to specify that the connection is via TLS.
>
> I tried setting "tls-bind-params" to a blank string but it didn't change
> anything. Is there any way to receive calls over TLS if you don't specify
> "transport=tls" in your contact string during registration?
>
> According to RFC3261 the use of the "transport=tls" parameter isn't
> recommended anymore and is now deprecated.
>
>
>   


From brian at freeswitch.org  Mon Dec  1 07:59:42 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2008 09:59:42 -0600
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <493406F4.204@gmx.net>
References: <20771637.post@talk.nabble.com> <493406F4.204@gmx.net>
Message-ID: <D6CA81AB-3B46-4A33-8076-18838D3392AA@freeswitch.org>

sip_secure_media only activates SRTP.

/b

On Dec 1, 2008, at 9:47 AM, Peter P GMX wrote:

> Did you add
> <action application="export" data="sip_secure_media=true"/>
> into youy dialplan before bridging that call. How is your  
> internal.conf,
> is TLS enabled there?
>
> Best regards
> Peter



From brian at freeswitch.org  Mon Dec  1 08:00:25 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2008 10:00:25 -0600
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <20771637.post@talk.nabble.com>
References: <20771637.post@talk.nabble.com>
Message-ID: <EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>

Please tell that to everyone out there in the REAL world.  It was my  
understanding that sips: was the one that went away in favor of  
transport= which is what everyone uses.

/b

On Dec 1, 2008, at 7:09 AM, matrim wrote:

> According to RFC3261 the use of the "transport=tls" parameter isn't
> recommended anymore and is now deprecated.



From mkarp at securesilence.com  Mon Dec  1 08:24:04 2008
From: mkarp at securesilence.com (Maxim Karp)
Date: Mon, 1 Dec 2008 08:24:04 -0800
Subject: [Freeswitch-users] Inbound 1-way audio issue using GSM codec
In-Reply-To: <4933CCDA.40905@gmx.net>
References: <49305CBF.8060801@ieee.org> <000301c951ba$9705dac0$c5119040$@com>
	<4933CCDA.40905@gmx.net>
Message-ID: <003e01c953d1$3b6f25e0$b24d71a0$@com>

Hi Peter,

Thanks for your response.

When I use PCMU two-way audio works fine.

When I make outgoing calls from a Freeswitch extension (using GSM) and then
out to a gateway using PCMU everything works fine.

When I receive calls from the same gateway, the end point behind the gateway
can't hear me.

The GSM-PSMU (and viceversa) transcoding for outgoing from an endpoint
associated with a  Freeswitch  extension to the external gateway is perfect
but incoming there seems to be an issue.

Maxim.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P
GMX
Sent: December-01-08 3:39 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inbound 1-way audio issue using GSM codec

Hello Maxim,

can you reach another internal device except the GSM one in order to see
whether it's GSM codec specific?

However I can see that you're using local IPs (10.x.x.x) so I expect
that they are natted. This often causes one way audio when the external
rtp-ip is not set. Please try to set a
    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
entry to internal.xml and external.xml in your SIP profiles and see if
it works. Use stun at least for the internal profile (FQDN and external
IP most probably will not work)

Best regards
Peter

Maxim Karp schrieb:
> Hello,
>
> I am using a GSM based endpoint connected to freeswitch that makes calls
to
> the PSTN via a SIP gateway (SBC).  The SBC uses PCMU between itself and
> freeswitch.
>
> When I make an outgoing call from a GSM based device via freewsitch to the
> PSTN via the SBC, everything works fine and audio works in both directions
> for both end points.  I looked at the console logs and they do indicate
that
> I am using GSM.
>
> Console output when I dial and before answer on the GSM device:
>
> v=0
> o=- 74 0 IN IP4 10.229.0.58
> s=session
> c=IN IP4 10.229.0.58
> b=CT:17
> t=0 0
> m=audio 59806 RTP/AVP 8 0 3 97 101
> k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 RED/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=encryption:optional
>
> Console output once it rings and after I answer on the PSTN side:
>
> v=0
> o=FreeSWITCH 1227887572 1227887573 IN IP4 10.229.0.10
> s=FreeSWITCH
> c=IN IP4 10.229.0.10
> t=0 0
> a=sendrecv
> m=audio 30896 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> When I receive a call from the SIP gateway, the endpoint making the call
> (not on freeswitch) can't hear me speaking from the GSM device connected
to
> freeswitch.  I can hear everything fine on the GSM device.
>
> Here is the console output for the call info coming in from the PSTN.
>
> v=0
> o=FreeSWITCH 1227902084 1227902085 IN IP4 38.113.164.132
> s=FreeSWITCH
> c=IN IP4 38.113.164.132
> t=0 0
> a=sendrecv
> m=audio 16724 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> Here is how I have vars.xml configured:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
>
> When I prioritize GSM on the outbound codec prefs I get static on the PSTN
> side. 
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/>
>
> Any ideas?
>
> Maxim.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Dec  1 08:26:27 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 10:26:27 -0600
Subject: [Freeswitch-users] Multi FS behind same NAT,
	PRACK goes to wrong port
In-Reply-To: <20081128142036.4190.qmail@server15.citromail.hu>
References: <20081128142036.4190.qmail@server15.citromail.hu>
Message-ID: <191c3a030812010826y20d60707o2b2a30973fbd7e11@mail.gmail.com>

if you enable nat mode on the registrations it will lock the ip:port
make an acl that matches the ip of the client  and add the param
apply-nat-acl with the name of the acl you created to your sofia profile
then all calls from that ip will be known to be nat and the port locking
code will activate.



On Fri, Nov 28, 2008 at 8:20 AM, x y <fs_ask_sy at citromail.hu> wrote:

> Hy!
>
> There are two different FS behind the same NAT, and there were Reigstration
> Failures about one or to times a day. The gateway status turned down, then I
> got 503 error codes. Then I set up the ext-ip to STUN, as the wiki requests
> it.
> Now I facing the next problem:
> Start the call, all goes right, INVITE goes to port 1352, then after 183
> Session progress from port 1352, the PRACK package goes to 5060 instead of
> 1352, wich messes up the call procedure. Is there anyway to force PRACK to
> the port to the INVITE has been sent before?
>
> Cheers,
> Viktor
>
>
>
> *Hirdet?s (x) *
> V?ltson most olcs?bb k?telez?re a biztos?t?s-hu-val. www.biztositas.hu<http://ad.adverticum.net/b/cl,1,2201,295040,360471/click.prm>- a k?telez? biztos?t?sok kiindul?pontja!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec  1 08:27:38 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 10:27:38 -0600
Subject: [Freeswitch-users] Set variable for the outgoing leg
In-Reply-To: <4933EE44.60900@kinetix.gr>
References: <4933EE44.60900@kinetix.gr>
Message-ID: <191c3a030812010827r64f6c66er8d3a5e49868430d6@mail.gmail.com>

if you use the export instead of set app then they will get set on both
legs.
otherwise vars you only want set on b leg you can add to the dial string
{foo=bar,test=true}sofia/default/user at dest.com


On Mon, Dec 1, 2008 at 8:01 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:

> All the variables that I set show up only in the a-leg CDR.
> How can I set a variable that can be used during the b-leg CDR generation?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec  1 08:29:11 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 10:29:11 -0600
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <586215.20730.qm@web30701.mail.mud.yahoo.com>
References: <586215.20730.qm@web30701.mail.mud.yahoo.com>
Message-ID: <191c3a030812010829v5860336ah35960774ee5d5af1@mail.gmail.com>

404 not found means the extension you are dialing is not found, not the
sound file the extension is playing.
press f8 and try again and the debug log will help you figure it out.


On Mon, Dec 1, 2008 at 12:09 AM, Faisal Maqsoodi
<faisalmaqsoodi at yahoo.com>wrote:

> I tried to play a sound file using the dialplan given on the link
> http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav
>
> <action application="playback" data="/path/to/your.wav"/>
> In place of /path/to/your.wave I used
> "/en/us/callie/misc/8000/call_secured.wav"
> "/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
> "/sounds/en/us/callie/misc/8000/call_secured.wav"
> But none of these is useful bcoz when i call on 2009, which is
> to b dialed to play the sound, same msg is
> displayed "404 NOT FOUND"
> Plz help me out.    Faisal
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ttroy50 at gmail.com  Mon Dec  1 08:42:30 2008
From: ttroy50 at gmail.com (Thomas Troy)
Date: Mon, 1 Dec 2008 16:42:30 +0000
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <493406F4.204@gmx.net>
References: <20771637.post@talk.nabble.com> <493406F4.204@gmx.net>
Message-ID: <adbe60280812010842t21283046ndd0d0734844eb738@mail.gmail.com>

I don't have that set however I'm not trying to use SRTP yet. At the moment
I'm just trying to use Secure SIP.

That section of my dial plan is

      <condition field="${sip_has_crypto}"
expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
break="never">
        <action application="set" data="sip_secure_media=true"/>
        <!-- Offer SRTP on outbound legs if we have it on inbound. -->
        <!-- <action application="export" data="sip_secure_media=true"/> -->
      </condition>

The TLS part of my internal.xml is now

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="true"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value=""/>
    <!-- Port to listen on for TLS requests. (5061 will be used if
unspecified) -->
    <param name="tls-sip-port" value="$${internal_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed
for TLS server) -->
    <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>


I also tried with

    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>

On Mon, Dec 1, 2008 at 3:47 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Did you add
> <action application="export" data="sip_secure_media=true"/>
> into youy dialplan before bridging that call. How is your internal.conf,
> is TLS enabled there?
>
> Best regards
> Peter
>
> matrim schrieb:
> > Hi,
> >
> > I'm having problems using TLS to receive calls.
> >
> > I'm using a Nokia N95 to test TLS against freeswitch. I can register my
> > client against freeswitch and make outbound calls to the test numbers
> (e.g.
> > 9999).
> >
> > I can also make calls to other users registered over UDP.
> >
> > However if I try to make a call to a user registered over TLS the leg of
> the
> > call to that user always goes via UDP.
> >
> > e.g.
> >
> > 1000 registered via TLS
> > 1001 registered via TLS
> > 1002 registered via UDP
> > 1003 registered via UDP
> >
> > 1000 -> 1002 works ok
> > 1003 -> 1002 works ok
> >
> > 1001 -> 1000 Doesn't work. The leg of the call between freeswitch and
> 1000
> > tries to setup via UDP
> > 1002 -> 1000 Doesn't work. The leg of the call between freeswitch and
> 1000
> > tries to setup via UDP
> >
> > ===
> >
> >
> >> >From looking at some of the documentation it seems to me that the issue
> may
> >>
> > be with the "tls-bind-params" being "transport=tls".
> >
> > The phone I'm using doesn't add the "transport=tls" parameter, and only
> uses
> > "sips:" to specify that the connection is via TLS.
> >
> > I tried setting "tls-bind-params" to a blank string but it didn't change
> > anything. Is there any way to receive calls over TLS if you don't specify
> > "transport=tls" in your contact string during registration?
> >
> > According to RFC3261 the use of the "transport=tls" parameter isn't
> > recommended anymore and is now deprecated.
> >
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ttroy50 at gmail.com  Mon Dec  1 08:49:43 2008
From: ttroy50 at gmail.com (Thomas Troy)
Date: Mon, 1 Dec 2008 16:49:43 +0000
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
Message-ID: <adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>

I'm not sure about current implementations that servers are using. I'm used
to using sip over UDP and TCP but this is my first time testing SIP over
TLS. So I'm just going by what's in the specification and what's implemented
on the devices I'm trying to test against, which are Nokia S60 devices (e.g.
Nokia N95, E66).

Out of interest do you have any links to anywhere this is discussed in terms
of general sip implementations?

On Mon, Dec 1, 2008 at 4:00 PM, Brian West <brian at freeswitch.org> wrote:

> Please tell that to everyone out there in the REAL world.  It was my
> understanding that sips: was the one that went away in favor of
> transport= which is what everyone uses.
>
> /b
>
> On Dec 1, 2008, at 7:09 AM, matrim wrote:
>
> > According to RFC3261 the use of the "transport=tls" parameter isn't
> > recommended anymore and is now deprecated.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Dec  1 09:02:08 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 11:02:08 -0600
Subject: [Freeswitch-users] Inbound 1-way audio issue using GSM codec
In-Reply-To: <000301c951ba$9705dac0$c5119040$@com>
References: <49305CBF.8060801@ieee.org> <000301c951ba$9705dac0$c5119040$@com>
Message-ID: <191c3a030812010902y688f6f08x5adfdd34349d4fde@mail.gmail.com>

probably pstn side has acknowledged our gsm then sent ulaw anyway and we
think its gsm.
most likely there are multiple codecs in the accept packet from the gateway
and they expect us to figure out what codec to use based on the first packet
we get from them rather than just accepting one codec in the sdp like 90% of
devices so we have a proper chance to setup optimal packetization.  This is
one of those lame parts of the RFC that describe complete unscalable
stupidity that some stuff likes to tout for who knows why.

one thing you can try is to set the variable aboslute_codec_string in the
dial to force
only gsm to be advertised at all making it impossible for the remote end to
respond with multiple codecs.

<action application="bridge"
data="{absolute_codec_string=GSM}sofia/<profile>/<uri>"/>




On Fri, Nov 28, 2008 at 6:36 PM, Maxim Karp <mkarp at securesilence.com> wrote:

> Hello,
>
> I am using a GSM based endpoint connected to freeswitch that makes calls to
> the PSTN via a SIP gateway (SBC).  The SBC uses PCMU between itself and
> freeswitch.
>
> When I make an outgoing call from a GSM based device via freewsitch to the
> PSTN via the SBC, everything works fine and audio works in both directions
> for both end points.  I looked at the console logs and they do indicate
> that
> I am using GSM.
>
> Console output when I dial and before answer on the GSM device:
>
> v=0
> o=- 74 0 IN IP4 10.229.0.58
> s=session
> c=IN IP4 10.229.0.58
> b=CT:17
> t=0 0
> m=audio 59806 RTP/AVP 8 0 3 97 101
> k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 RED/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=encryption:optional
>
> Console output once it rings and after I answer on the PSTN side:
>
> v=0
> o=FreeSWITCH 1227887572 1227887573 IN IP4 10.229.0.10
> s=FreeSWITCH
> c=IN IP4 10.229.0.10
> t=0 0
> a=sendrecv
> m=audio 30896 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> When I receive a call from the SIP gateway, the endpoint making the call
> (not on freeswitch) can't hear me speaking from the GSM device connected to
> freeswitch.  I can hear everything fine on the GSM device.
>
> Here is the console output for the call info coming in from the PSTN.
>
> v=0
> o=FreeSWITCH 1227902084 1227902085 IN IP4 38.113.164.132
> s=FreeSWITCH
> c=IN IP4 38.113.164.132
> t=0 0
> a=sendrecv
> m=audio 16724 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> Here is how I have vars.xml configured:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
>
> When I prioritize GSM on the outbound codec prefs I get static on the PSTN
> side.
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/>
>
> Any ideas?
>
> Maxim.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec  1 09:06:56 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 11:06:56 -0600
Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 29, Issue 189
In-Reply-To: <f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>
References: <mailman.1445.1228105114.18013.freeswitch-users@lists.freeswitch.org>
	<f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>
Message-ID: <191c3a030812010906m8ed33b2i6a9d65dc8b1c962f@mail.gmail.com>

we no longer use global name space in our shared objects which seems to have
a side effect on modules who in turn try to load it's own shared objects
because they too inherit the non-global namespace param.

you can either add an attribute to the modues.conf to ask it to load with
global name space

<load module="mod_foo" global="true"/>

or you can edit the code to request global loading every time.

by adding the param SMODF_GLOBAL_SYMBOLS to the SWITCH_MODULE_DEFINITION
macro
(See mod_cepstral at the top)


On Mon, Dec 1, 2008 at 2:31 AM, Traun Leyden <tleyden at branchcut.com> wrote:

>
>
>> Message: 9
>> Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST)
>> From: Marc Orenberg <marc at kasteris.com>
>> Subject: [Freeswitch-users] Problem importing modules in mod_python
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID: <195670.44941.qm at web50805.mail.re2.yahoo.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> In the latter versions of mod_python, I'm unable to import standard python
>> modules such as time and MySQLdb.?
>> For example, the following script works fine in version 1.0.1:
>>
>> ??? ??? import time
>> ??? ??? import os
>> ??? ??? from freeswitch import *
>> ??? ??? def handler(session, args):
>> ??? ??? ??? session.answer()
>> ??? ??? ??? session.execute("sleep", "2000")
>> ??? ??? ???
>> session.streamFile("/usr/local/freeswitch/prompts/01Welcome.wav")
>> ??? ??? ??? return(session)
>>
>> But in freeswitch-1.0.latest.tar.gz, and svn versions 10556-10558, I get
>> the following error:
>>
>> ??? ??? 2008-11-30 21:13:09 [ERR] mod_python.c:129 eval_some_python()
>> Error reloading module
>> ??? ??? Traceback (most recent call last):
>> ??????? File "/usr/lib/python2.4/site-packages/scripts/test.py", line 1,
>> in ?
>> ??? ??? import time
>> ??? ??? ImportError: /usr/lib/python2.4/lib-dynload/timemodule.so:
>> undefined symbol: PyExc_ValueError
>>
>> Thanks for your help!
>>
>
> I have run into the same problem and put some documentation on the wiki:
>
>
> http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2F...2Fdatetime.so:_undefined_symbol:_PyExc_IOError
>
> I think something changed in freeswitch in the way it is loading modules,
> or at least the way it is loading mod_python.  This behavior appeared all of
> the sudden in recent freeswitch versions.
>
> HTH
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec  1 09:29:00 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 11:29:00 -0600
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
Message-ID: <191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>

yes,

mod_shout will broadcast calls as MP3 that you can listen to in
itunes/winamp live.


On Fri, Nov 28, 2008 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:

> so i would have to make a call with a phone to a specific dialplan? if
> so, this would not be, what i whished (although it is nice to have the
> option).
>
> isn't there something, which can stream the voice of a given uuid? so
> i could place a link in the html admin-area to spy an uuid and to hear
> everything over the speaker? this would be really sexy ;)
>
>
> 2008/11/18 Anthony Minessale <anthony.minessale at gmail.com>:
> > you can use the eavesdrop dialplan app from a new call to spy on an in
> > progress session
> > it takes the uuid of the channel you want to listen to as the arg.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Mon Dec  1 09:28:34 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Dec 2008 12:28:34 -0500
Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 29, Issue 189
In-Reply-To: <f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>
References: <mailman.1445.1228105114.18013.freeswitch-users@lists.freeswitch.org>
	<f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>
Message-ID: <8DC3637C-9C3D-4F25-9361-06165893F116@jerris.com>

Try changing the module definition to use global symbols the same way  
we did in mod_lua, see if that resolves the issue.

Mike

On Dec 1, 2008, at 3:31 AM, Traun Leyden wrote:

>
>
> Message: 9
> Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST)
> From: Marc Orenberg <marc at kasteris.com>
> Subject: [Freeswitch-users] Problem importing modules in mod_python
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <195670.44941.qm at web50805.mail.re2.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> In the latter versions of mod_python, I'm unable to import standard  
> python modules such as time and MySQLdb.?
> For example, the following script works fine in version 1.0.1:
>
> ??? ??? import time
> ??? ??? import os
> ??? ??? from freeswitch import *
> ??? ??? def handler(session, args):
> ??? ??? ??? session.answer()
> ??? ??? ??? session.execute("sleep", "2000")
> ??? ??? ??? session.streamFile("/usr/local/freeswitch/prompts/ 
> 01Welcome.wav")
> ??? ??? ??? return(session)
>
> But in freeswitch-1.0.latest.tar.gz, and svn versions 10556-10558, I  
> get the following error:
>
> ??? ??? 2008-11-30 21:13:09 [ERR] mod_python.c:129  
> eval_some_python() Error reloading module
> ??? ??? Traceback (most recent call last):
> ??????? File "/usr/lib/python2.4/site-packages/scripts/test.py",  
> line 1, in ?
> ??? ??? import time
> ??? ??? ImportError: /usr/lib/python2.4/lib-dynload/timemodule.so:  
> undefined symbol: PyExc_ValueError
>
> Thanks for your help!
>
> I have run into the same problem and put some documentation on the  
> wiki:
>
> http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2F... 
> 2Fdatetime.so:_undefined_symbol:_PyExc_IOError
>
> I think something changed in freeswitch in the way it is loading  
> modules, or at least the way it is loading mod_python.  This  
> behavior appeared all of the sudden in recent freeswitch versions.
>
> HTH
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Dec  1 09:37:45 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 11:37:45 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
Message-ID: <191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>

the guy who made mod_openmrcp has stopped development and is now making a
new library called
unimrcp it will take some time to create a new module and remove the now
unsupported openmrcp.



On Fri, Nov 28, 2008 at 12:15 PM, <mszlazak at aol.com> wrote:

> I'm getting the following errors when trying to run the example in the
> wiki: http://wiki.freeswitch.org/wiki/Mod_openmrcp
>
> 2008-11-28 09:59:54 [DEBUG] switch_core_session.c:435
> switch_core_session_receive_message() Send signal sofia/internal/
> 1000 at 10.0.0.2 [BREAK]
> 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel
> sofia/internal/1000 at 10.0.0.2 entering state [completed]
> 2008-11-28 09:59:54 [NOTICE] mod_spidermonkey.c:2034 session_answer()
> Channel [sofia/internal/1000 at 10.0.0.2] has been answered
> 2008-11-28 09:59:54 [DEBUG] mod_spidermonkey.c:1851 init_speech_engine()
> Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
> 2008-11-28 09:59:54 [DEBUG] mod_openmrcp.c:634 openmrcp_tts_open() Create
> Synthesizer Channel
> 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel
> sofia/internal/1000 at 10.0.0.2 entering state [ready]
>
> 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:643 openmrcp_tts_open() No
> response from client stack
> 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:647 openmrcp_tts_open() No
> synthesizer channel available
> 2008-11-28 09:59:59 [ERR] mod_spidermonkey.c:1859 init_speech_engine()
> Invalid TTS module!
> 2008-11-28 09:59:59 [ERR] inline:1 mod_spidermonkey()  Cannot allocate
> speech engine!
>
> 2008-11-28 09:59:59 [NOTICE] switch_core_state_machine.c:160
> switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2[CS_EXECUTE] [NORMAL_CLEARING]
> 2008-11-28 09:59:59 [DEBUG] switch_channel.c:1449
> switch_channel_perform_hangup() Send signal sofia/internal/1000 at 10.0.0.2[KILL]
> 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:727
> switch_core_session_signal_state_change() Send signal sofia/internal/
> 1000 at 10.0.0.2 [BREAK]
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:432
> switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State EXECUTE
> going to sleep
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:367
> switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) Running State
> Change CS_HANGUP
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395
> switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State HANGUP
> 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:276 sofia_on_hangup() Channel
> sofia/internal/1000 at 10.0.0.2 hanging up, cause: NORMAL_CLEARING
> 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:333 sofia_on_hangup() Sending BYE
> to sofia/internal/1000 at 10.0.0.2
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:46
> switch_core_standard_on_hangup() sofia/internal/1000 at 10.0.0.2 Standard
> HANGUP, cause: NORMAL_CLEARING
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395
> switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State HANGUP
> going to sleep
> 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:860
> switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2)
> Locked, Waiting on external entities
> 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:878
> switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2)
> Ended
> 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:880
> switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2[CS_HANGUP]
> 2008-11-28 10:00:26 [DEBUG] mod_openmrcp.c:167
> openmrcp_on_session_terminate() on_session_terminate called
>
> I believe I followed the instructions correctly but I can't get openmrcp to
> connect with Cepstrals TTS.
>
> ------------------------------
> Tis the season to save your money! Get the new AOL Holiday Toolbar<http://toolbar.aol.com/holiday/download.html?ncid=emlweusdown00000008>for money saving offers and gift ideas.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From simon at airg.com  Mon Dec  1 10:01:47 2008
From: simon at airg.com (Simon Tang)
Date: Mon, 1 Dec 2008 10:01:47 -0800
Subject: [Freeswitch-users] Leg A terminated by Leg B on a uuid_bridge
In-Reply-To: <5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>
References: <872970CF4A55BF42A5337D570860209F01052E34@HPEXCHVS01.exchange.airg>
	<5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>
Message-ID: <872970CF4A55BF42A5337D570860209F01052EA8@HPEXCHVS01.exchange.airg>

Thanks Dennis,

That did exactly what I needed.

Cheers!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Dennis
Sent: December 1, 2008 3:47 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Leg A terminated by Leg B on a
uuid_bridge

hi  simon,

i am not sure, if i understood your problem right, but if you do not
want leg a to hang up after leg b (the originated call) hangs up, set
"park_after_bridge=true" when you make the originate.
as far as i know, "hangup_after_bridge=false" is only for the inbound
and helps nothing with the outbound.

if you want something different, please explain me a little more.

dennis



2008/11/28 Simon Tang <simon at airg.com>:
> Hello,
>
>
>
> I'm using event socket outbound, and have an issue where, after a
bridge
> ends and is terminated by Leg B, Leg A is also terminated.  Here's the
call
> flow:
>
>
>
> 1.       Call comes in (Leg A), session created, play welcome message.
>
> 2.       From this session, originate and dial out using api originate
>
> 3.       After the target answers (Leg B), bridge the 2 calls using
api
> uuid_bridge
>
> 4.       Leg B hangs up.
>
> 5.       Leg A will be terminated.
>
>
>
> After step 4, Leg A is terminated.  I do not want Leg A to hang up.
I've
> tried setting "hangup_after_bridge=false" prior to the call, and that
> doesn't work.
>
>
>
> Having said that, I tried a similar test which does not end Leg A's
call
> after Leg B hangs up, but I can't use this solution because,
functionally,
> does not accomplish what I want it to do (i.e., I want to perform some
> actions on Leg B prior to the bridge, like send some DTMF tones,
playback
> some messages, etc).  I did not need to set the "hangup_after_bridge"
> variable (default should be false anyway).
>
>
>
> 1.       Call comes in (Leg A), session created, play welcome message.
>
> 2.       From this session, do a bridge by doing an execute bridge.
>
> 3.       The target answers (Leg B)
>
> 4.       Leg B hangs up.
>
> 5.       Leg A will still be active.
>
>
>
> Any ideas would be appreciated.  Thanks!
>
>
>
> Simon
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mszlazak at aol.com  Mon Dec  1 10:19:42 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 01 Dec 2008 13:19:42 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
Message-ID: <8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>


 Hi Anthony,

Oh! OK.

So is this module "totally broken". 

I say this because I can't seem to get it to work at all with the example in that Mod_openmrcp wiki page but I thought it might because I'm not be using the right Cepstral software (freetrial download versus the paided for SDK) or that I'm not using the right port numbers or something else I didn't do. I used TcpView to look at local port associated with my Cepstral software and changed a few things but still nothing. I changed the loglevel setting to 7 in the wiki's example but I don't see the kind of output on the console that I would expect for debug mode.

Thanks. Mark.


 


 

-----Original Message-----
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 1 Dec 2008 9:37 am
Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP









the guy who made mod_openmrcp has stopped development and is now making a new library called 
unimrcp it will take some time to create a new module and remove the now unsupported openmrcp.





On Fri, Nov 28, 2008 at 12:15 PM,  <mszlazak at aol.com> wrote:


I'm getting the following errors when trying to run the example in the wiki: http://wiki.freeswitch.org/wiki/Mod_openmrcp




2008-11-28 09:59:54 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Send signal sofia/internal/1000 at 10.0.0.2 [BREAK]

2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 10.0.0.2 entering state [completed]

2008-11-28 09:59:54 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000 at 10.0.0.2] has been answered

2008-11-28 09:59:54 [DEBUG] mod_spidermonkey.c:1851 init_speech_engine() Raw Codec Activation Success L16 at 8000hz 1 channel 20ms

2008-11-28 09:59:54 [DEBUG] mod_openmrcp.c:634 openmrcp_tts_open() Create Synthesizer Channel

2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 10.0.0.2 entering state [ready]



2008-11-28 09:59:59 [ERR] mod_openmrcp.c:643 openmrcp_tts_open() No response from client stack

2008-11-28 09:59:59 [ERR] mod_openmrcp.c:647 openmrcp_tts_open() No synthesizer channel available

2008-11-28 09:59:59 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!

2008-11-28 09:59:59 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine!



2008-11-28 09:59:59 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]

2008-11-28 09:59:59 [DEBUG] switch_channel.c:1449 switch_channel_perform_hangup() Send signal sofia/internal/1000 at 10.0.0.2 [KILL]

2008-11-28 09:59:59 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal sofia/internal/1000 at 10.0.0.2 [BREAK]

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State EXECUTE going to sleep

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:367 switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) Running State Change CS_HANGUP

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State HANGUP

2008-11-28 09:59:59 [DEBUG] mod_sofia.c:276 sofia_on_hangup() Channel sofia/internal/1000 at 10.0.0.2 hanging up, cause: NORMAL_CLEARING

2008-11-28 09:59:59 [DEBUG] mod_sofia.c:333 sofia_on_hangup() Sending BYE to sofia/internal/1000 at 10.0.0.2

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1000 at 10.0.0.2 Standard HANGUP, cause: NORMAL_CLEARING

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State HANGUP going to sleep

2008-11-28 09:59:59 [DEBUG] switch_core_session.c:860 switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2) Locked, Waiting on external entities

2008-11-28 09:59:59 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2) Ended

2008-11-28 09:59:59 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]

2008-11-28 10:00:26 [DEBUG] mod_openmrcp.c:167 openmrcp_on_session_terminate() on_session_terminate called



I believe I followed the instructions correctly but I can't get openmrcp to connect with Cepstrals TTS.






 Tis the season to save your money!  Get the new AOL Holiday Toolbar for money saving offers and gift ideas. 



_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org







-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/


AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sergey.kirillov at gmail.com  Mon Dec  1 07:26:58 2008
From: sergey.kirillov at gmail.com (Sergey Kirillov)
Date: Mon, 01 Dec 2008 17:26:58 +0200
Subject: [Freeswitch-users] Support for Junghanns duoBRI
Message-ID: <49340242.3040403@gmail.com>

Greetings,

Can somebody tell me, if it is possible to use duoBRI card
(http://www.junghanns.net/en/duobri_express_produkt.html) from
Junghanns.net together with Freeswitch?

I've found that this card has Zaptel drivers, and Freeswitch has
mod_openzap. On the other side, I saw somewhere in wiki that Freeswitch
does not support BRI at all at the moment.


Please confirm or allay my apprehensions.



From mike at jerris.com  Mon Dec  1 10:30:28 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Dec 2008 13:30:28 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
Message-ID: <CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>

I would not say it is totally broken, it is known to work in quite a  
few places, but we are unlikely to be doing any new fixes in it.

Mike

On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:

> Hi Anthony,
>
> Oh! OK.
>
> So is this module "totally broken".
>
> I say this because I can't seem to get it to work at all with the  
> example in that Mod_openmrcp wiki page but I thought it might  
> because I'm not be using the right Cepstral software (freetrial  
> download versus the paided for SDK) or that I'm not using the right  
> port numbers or something else I didn't do. I used TcpView to look  
> at local port associated with my Cepstral software and changed a few  
> things but still nothing. I changed the loglevel setting to 7 in the  
> wiki's example but I don't see the kind of output on the console  
> that I would expect for debug mode.
>
> Thanks. Mark.
>
>
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From mike at jerris.com  Mon Dec  1 10:31:32 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Dec 2008 13:31:32 -0500
Subject: [Freeswitch-users] Support for Junghanns duoBRI
In-Reply-To: <49340242.3040403@gmail.com>
References: <49340242.3040403@gmail.com>
Message-ID: <B22D2E3D-9C6F-43C6-A634-DC00B91BDBBB@jerris.com>

The bri support is still in development, basic calls on ptmp bri do  
appear to work, although I am not sure with what hardware.

Mike


On Dec 1, 2008, at 10:26 AM, Sergey Kirillov wrote:

> Greetings,
>
> Can somebody tell me, if it is possible to use duoBRI card
> (http://www.junghanns.net/en/duobri_express_produkt.html) from
> Junghanns.net together with Freeswitch?
>
> I've found that this card has Zaptel drivers, and Freeswitch has
> mod_openzap. On the other side, I saw somewhere in wiki that  
> Freeswitch
> does not support BRI at all at the moment.
>
>
> Please confirm or allay my apprehensions.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mszlazak at aol.com  Mon Dec  1 10:41:38 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 01 Dec 2008 13:41:38 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com><191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com><8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
Message-ID: <8CB21FB943565E5-CA4-C41@MBLK-M05.sysops.aol.com>


 

MikeJ, if openMRCP isn't totally broken then would you mind helping me get the example in Mod_openMRCP working or something like it since I don't know what the heck I'm doing wrong.

I can meet you now over at the IRC channel for Freeswitch users if you like.

Thanks.

 

-----Original Message-----
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 1 Dec 2008 10:30 am
Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP









I would not say it is totally broken, it is known to work in quite a few places, but we are unlikely to be doing any new fixes in it.



Mike




On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:



 Hi Anthony,
 
 Oh! OK.
 
 So is this module "totally broken". 
 
 I say this because I can't seem to get it to work at all with the example in that Mod_openmrcp wiki page but I thought it might because I'm not be using the right Cepstral software (freetrial download versus the paided for SDK) or that I'm not using the right port numbers or something else I didn't do. I used TcpView to look at local port associated with my Cepstral software and changed a few things but still nothing. I changed the loglevel setting to 7 in the wiki's example but I don't see the kind of output on the console that I would expect for debug mode.
 
 Thanks. Mark.
 
 
 
 
 





=


 





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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From dave at 3c.co.uk  Mon Dec  1 10:51:39 2008
From: dave at 3c.co.uk (David Knell)
Date: Mon, 01 Dec 2008 18:51:39 +0000
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
Message-ID: <4934323B.9000305@3c.co.uk>

Hi Mike,

My experience is that it's somewhat broken - it took two trivial tweaks 
to get it to work with IBM's ASR and TTS, but there's a more intractable 
problem to do with memory getting overwritten (I assume that this is 
something to do with something being freed when it shouldn't be) which 
causes a segfault on the second or third session after the module being 
loaded.

Without wishing to sound like a stuck record, one thing that you guys 
really ought to do is to decide what's supported and what isn't, and 
make this obvious - for example, move unsupported modules to a different 
place in the tree, don't have them built by default, etc.  MRCP is in 
the specsheet on the Wiki.  Otherwise folk like Mark and I spend time 
installing stuff, go round in circles a bit trying to make it work, and 
then find out (a) that it doesn't and (b) it's not going to be fixed 
because it's not supported.

Cheers --

Dave
> I would not say it is totally broken, it is known to work in quite a 
> few places, but we are unlikely to be doing any new fixes in it.
>
> Mike
>
> On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com <mailto:mszlazak at aol.com> 
> wrote:
>
>> Hi Anthony,
>>
>> Oh! OK.
>>
>> So is this module "totally broken".
>>
>> I say this because I can't seem to get it to work at all with the 
>> example in that Mod_openmrcp wiki page but I thought it might because 
>> I'm not be using the right Cepstral software (freetrial download 
>> versus the paided for SDK) or that I'm not using the right port 
>> numbers or something else I didn't do. I used TcpView to look at 
>> local port associated with my Cepstral software and changed a few 
>> things but still nothing. I changed the loglevel setting to 7 in the 
>> wiki's example but I don't see the kind of output on the console that 
>> I would expect for debug mode.
>>
>> Thanks. Mark.
>>
>>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From anthony.minessale at gmail.com  Mon Dec  1 11:17:56 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 13:17:56 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <4934323B.9000305@3c.co.uk>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
Message-ID: <191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>

mod_openmrcp was a contribution to the community by a 3rd party individual.

As i have clearly stated in 2 previous emails, the man has decided to
discontinue the openmrcp project.
So now we are left with the remains of the module and discontinued code.
This was not our decision it was his.

Since the author of openmrcp has stated that he has a new unimrcp we are
certainly going to
work towards getting mod_unimrcp to replace mod_openmrcp.  He had already
commented on that previous thread to state he is willing to consider making
a new module.

Some people use it without issue which may mean that the crash you reported
is windows specific and I do not have a working lab of any mrcp capbable
system to try it against in unix for that matter.  I have a list of work to
do from here to the moon and back so on an issue like this, unless someone
can hand me login credentials to some box and give me a phone number to dial
to reporduce the issue, it will be a long time until we can deal with it.
And the question arises, should we bother working on it anymore if the lib
has been abandoned and we cannot even get any support from it's author which
is where the problem most likely lies.

I try not to get too annoyed by these remarks about what we *ought to do*
because I know people lose sight of how much of the work to support the
project is done by a small group of 3 people and not the 2000 people it
appears to be from the outside looking in. (I've been answering email for 4
hours now)

My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
FS that we can maintain in tree knowing the development can be supported by
the original author.


On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:

>  Hi Mike,
>
> My experience is that it's somewhat broken - it took two trivial tweaks to
> get it to work with IBM's ASR and TTS, but there's a more intractable
> problem to do with memory getting overwritten (I assume that this is
> something to do with something being freed when it shouldn't be) which
> causes a segfault on the second or third session after the module being
> loaded.
>
> Without wishing to sound like a stuck record, one thing that you guys
> really ought to do is to decide what's supported and what isn't, and make
> this obvious - for example, move unsupported modules to a different place in
> the tree, don't have them built by default, etc.  MRCP is in the specsheet
> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
> round in circles a bit trying to make it work, and then find out (a) that it
> doesn't and (b) it's not going to be fixed because it's not supported.
>
> Cheers --
>
> Dave
>
> I would not say it is totally broken, it is known to work in quite a few
> places, but we are unlikely to be doing any new fixes in it.
>  Mike
>
>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>
>  Hi Anthony,
>
> Oh! OK.
>
> So is this module "totally broken".
>
> I say this because I can't seem to get it to work at all with the example
> in that Mod_openmrcp wiki page but I thought it might because I'm not be
> using the right Cepstral software (freetrial download versus the paided for
> SDK) or that I'm not using the right port numbers or something else I didn't
> do. I used TcpView to look at local port associated with my Cepstral
> software and changed a few things but still nothing. I changed the loglevel
> setting to 7 in the wiki's example but I don't see the kind of output on the
> console that I would expect for debug mode.
>
> Thanks. Mark.
>
>
>   ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> David Knell, Director, 3C Limited
> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gkuri at ieee.org  Mon Dec  1 11:34:03 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 01 Dec 2008 11:34:03 -0800
Subject: [Freeswitch-users] SIP INVITE timeout
In-Reply-To: <8DE97AFC-64E1-45A4-9B33-21C6300F52B4@freeswitch.org>
References: <49305CBF.8060801@ieee.org>
	<8DE97AFC-64E1-45A4-9B33-21C6300F52B4@freeswitch.org>
Message-ID: <49343C2B.3020302@ieee.org>

Brian,

Will setting progress_timeout = 8 and originate_timeout = 30 help me out
in this situation without using pre_answer?

Basically I'd like to timeout the INVITE to the phone in 8 seconds if it
doesn't respond to the INVITE (phone is not on the network) and send the
call to voicemail, but if the phone is actually ringing and no one picks
up in 30 seconds, send it to voicemail?

Thanks
Gabe

Brian West wrote:
> Try pre_answer before bridge.
> 
> /b
> 
> Sent from my iPhone
> 
> On Nov 28, 2008, at 3:03 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> 
>> I have a phone that is registered to FS but is no longer available
>> (Internet connection down, phone turned off, etc.). The registration
>> still exists in the sip_registrations table (not expired yet), but the
>> phone is not reachable on the network.
>>
>> According to my dialplan, if the bridge to the phone fails after 20
>> seconds, the call should be forwarded to a different box for handling
>> (see dialplan below).
>>
>>
>>      <extension name="My Extension">
>>       <condition field="destination_number" expression="1213XXXXXXX">
>>             <action application="info"/>
>>        <action application="set" data="originate_timeout=20"/>
>>             <action application="set" 
>> data="hangup_after_bridge=true"/>
>>             <action application="bridge"
>> data="${sofia_contact(default/1213XXXXXXX at mydomain.net"/>
>>        <action application="bridge"
>> data="sofia/default/1213XXXXXXX at box.mydomain.net"/>
>>        </condition>
>>      </extension>
>>
>>
>> If the phone is down and not responding to the INVITEs, it appears my
>> carrier is canceling the SIP INVITE to FreeSWITCH after about 10
>> seconds. My timeout is 20 seconds. Is there anyway to deal with this
>> situation, without going back to my carrier and asking them to 
>> increase
>> their timeout on an INVITE?
>>
>> Call Progress:
>>
>> Carrier -> FS (INVITE)
>> FS -> Carrier (100 Trying)
>>
>> <10 seconds pass while FS is attempting to contact the phone>
>>
>> Carrier -> FS (CANCEL)
>> FS -> Carrier (200 OK)
>> FS -> Carrier (487 Request Terminated)
>> Carrier -> FS (ACK)
>>
>>
>> Thanks ...
>>
>> Gabe
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From msc at freeswitch.org  Mon Dec  1 11:37:27 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Dec 2008 11:37:27 -0800
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
Message-ID: <87f2f3b90812011137v6f8c9125x9b2ae7b7f5e5bc21@mail.gmail.com>

FYI,
I've updated the wiki to reflect the current status of OpenMRCP with a link
to the new UniMRCP project. Hopefully enough people who want MRCP in FS will
support UniMRCP...
-MC

On Mon, Dec 1, 2008 at 11:17 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> mod_openmrcp was a contribution to the community by a 3rd party individual.
>
> As i have clearly stated in 2 previous emails, the man has decided to
> discontinue the openmrcp project.
> So now we are left with the remains of the module and discontinued code.
> This was not our decision it was his.
>
> Since the author of openmrcp has stated that he has a new unimrcp we are
> certainly going to
> work towards getting mod_unimrcp to replace mod_openmrcp.  He had already
> commented on that previous thread to state he is willing to consider making
> a new module.
>
> Some people use it without issue which may mean that the crash you reported
> is windows specific and I do not have a working lab of any mrcp capbable
> system to try it against in unix for that matter.  I have a list of work to
> do from here to the moon and back so on an issue like this, unless someone
> can hand me login credentials to some box and give me a phone number to dial
> to reporduce the issue, it will be a long time until we can deal with it.
> And the question arises, should we bother working on it anymore if the lib
> has been abandoned and we cannot even get any support from it's author which
> is where the problem most likely lies.
>
> I try not to get too annoyed by these remarks about what we *ought to do*
> because I know people lose sight of how much of the work to support the
> project is done by a small group of 3 people and not the 2000 people it
> appears to be from the outside looking in. (I've been answering email for 4
> hours now)
>
> My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
> FS that we can maintain in tree knowing the development can be supported by
> the original author.
>
>
> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>
>>  Hi Mike,
>>
>> My experience is that it's somewhat broken - it took two trivial tweaks to
>> get it to work with IBM's ASR and TTS, but there's a more intractable
>> problem to do with memory getting overwritten (I assume that this is
>> something to do with something being freed when it shouldn't be) which
>> causes a segfault on the second or third session after the module being
>> loaded.
>>
>> Without wishing to sound like a stuck record, one thing that you guys
>> really ought to do is to decide what's supported and what isn't, and make
>> this obvious - for example, move unsupported modules to a different place in
>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>> round in circles a bit trying to make it work, and then find out (a) that it
>> doesn't and (b) it's not going to be fixed because it's not supported.
>>
>> Cheers --
>>
>> Dave
>>
>> I would not say it is totally broken, it is known to work in quite a few
>> places, but we are unlikely to be doing any new fixes in it.
>>  Mike
>>
>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>
>>  Hi Anthony,
>>
>> Oh! OK.
>>
>> So is this module "totally broken".
>>
>> I say this because I can't seem to get it to work at all with the example
>> in that Mod_openmrcp wiki page but I thought it might because I'm not be
>> using the right Cepstral software (freetrial download versus the paided for
>> SDK) or that I'm not using the right port numbers or something else I didn't
>> do. I used TcpView to look at local port associated with my Cepstral
>> software and changed a few things but still nothing. I changed the loglevel
>> setting to 7 in the wiki's example but I don't see the kind of output on the
>> console that I would expect for debug mode.
>>
>> Thanks. Mark.
>>
>>
>>   ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.kubr at gmail.com  Mon Dec  1 12:48:53 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Mon, 1 Dec 2008 21:48:53 +0100
Subject: [Freeswitch-users] Sound file as ringback
Message-ID: <698401620812011248g53ef6579q5ca03ba22ce69709@mail.gmail.com>

Yes, setting the var to the full path works. Sorry, should have taken
the "full path" in the wiki more seriously.
MP3s are played only once, 8kHz WAVs work perfectly.

Cheers,
Jan

> Can you try putting the full path to the file?  Also what does the
> console output look like?
>
> /b
>
> On Nov 30, 2008, at 12:30 PM, Jan Kubr wrote:
>
>> I have try different format of files (from 8KHz mono wavs to MP3s, all
>> of which play fine via playback) and some caused the bridge to be
>> finished immediately (with NO_USER_RESPONSE), some make it generate
>> crazy beeping, but none is played while the phone is ringing on the
>> other end.


From jan.kubr at gmail.com  Mon Dec  1 12:55:51 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Mon, 1 Dec 2008 21:55:51 +0100
Subject: [Freeswitch-users] How to specify Path for sound files
Message-ID: <698401620812011255g2e15991w4970559ac912f9b3@mail.gmail.com>

Hi Faisal,
the path is either an absolute path or a path relative to the
directory in the sound_prefix var in vars.xml.
So this

<action application="playback"
data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"/>

works fine on my box. You sure this one doesn't work for you?

Jan

> I tried to play a sound file using the dialplan given on the link
> http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav
>
> <action application="playback" data="/path/to/your.wav"/>
> In place of /path/to/your.wave I used
> "/en/us/callie/misc/8000/call_secured.wav"
> "/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
> "/sounds/en/us/callie/misc/8000/call_secured.wav"
> But none of these is useful bcoz when i call on 2009, which is
> to b dialed to play the sound, same msg is
> displayed "404 NOT FOUND"
> Plz help me out.??? Faisal


From mszlazak at aol.com  Mon Dec  1 16:44:31 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 01 Dec 2008 19:44:31 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com><191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com><8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com><CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com><4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
Message-ID: <8CB222E46498CA1-11E0-E8B@MBLK-M05.sysops.aol.com>


 


 Does "bridging" a call from FS to Voxeo's Prophecy server require openMRCP? If not then the other issue I might have is a database look up that is part of the dialogue that maybe need as the person response to prompts from the asr. It's possible to run a php script for the database stuff that Prophecy might need or could that happen via Javascript in FS? Then after the dialogue has completed I go from Prophecy back to FS.


 

-----Original Message-----
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 1 Dec 2008 11:17 am
Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP









mod_openmrcp was a contribution to the community by a 3rd party individual.

As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project.
So now we are left with the remains of the module and discontinued code.? This was not our decision it was his. 


Since the author of openmrcp has stated that he has a new unimrcp we are certainly going to 
work towards getting mod_unimrcp to replace mod_openmrcp.? He had already commented on that previous thread to state he is willing to consider making a new module.



Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter.? I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it.? And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies.


I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) 


My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author.



On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:





  



Hi Mike,



My experience is that it's somewhat broken - it took two trivial tweaks
to get it to work with IBM's ASR and TTS, but there's a more
intractable problem to do with memory getting overwritten (I assume
that this is something to do with something being freed when it
shouldn't be) which causes a segfault on the second or third session
after the module being loaded.



Without wishing to sound like a stuck record, one thing that you guys
really ought to do is to decide what's supported and what isn't, and
make this obvious - for example, move unsupported modules to a
different place in the tree, don't have them built by default, etc.?
MRCP is in the specsheet on the Wiki.? Otherwise folk like Mark and I
spend time installing stuff, go round in circles a bit trying to make
it work, and then find out (a) that it doesn't and (b) it's not going
to be fixed because it's not supported.



Cheers --



Dave






I would not say it is totally broken, it is known to work
in quite a few places, but we are unlikely to be doing any new fixes in
it.
  


  

  
Mike

  


  

  
On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:

  

  

    
 Hi Anthony,

    

Oh! OK.

    

So is this module "totally broken". 

    

I say this because I can't seem to get it to work at all with the
example in that Mod_openmrcp wiki page but I thought it might because
I'm not be using the right Cepstral software (freetrial download versus
the paided for SDK) or that I'm not using the right port numbers or
something else I didn't do. I used TcpView to look at local port
associated with my Cepstral software and changed a few things but still
nothing. I changed the loglevel setting to 7 in the wiki's example but
I don't see the kind of output on the console that I would expect for
debug mode.

    

Thanks. Mark.

    

    
 

    

    


    

  

  

  

  






_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  









-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 





_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org







-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/


AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From kkielhofner at star2star.com  Mon Dec  1 19:43:01 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Mon, 1 Dec 2008 22:43:01 -0500
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
Message-ID: <2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>

On 12/1/08, Thomas Troy <ttroy50 at gmail.com> wrote:
..snip..
>
> Out of interest do you have any links to anywhere this is discussed in terms
> of general sip implementations?
>

Uh oh, here we go again...

http://www.iana.org/assignments/sip-parameters
http://tools.ietf.org/html/rfc3969

https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-August/010047.html

Implementation wise, most devices tend to use transport=tls:

SIPFoundry - From what I've seen
Snom
SERs
Asterisk (If you are using TLS)
Cisco - I *believe* you can use either a SIPS URI or the transport=tls
parameter for various SIP targets

  As the RFC (basically) states (RFC3261, section 12.1.x),
transport=tls was deprecated in RFC 3261 because you should also be
able to do TLS over SCTP (RFC3436), which makes transport=tls a bit
ambiguous. sips:user at domain;transport=tcp or
sips:user at domain;transport=sctp is a bit more flexible.

  I don't know if I've ever seen anything default to SIPS URIs.  I
also don't think I've ever specifically tried using them.  However, my
experience with TLS is admittedly somewhat limited so this shouldn't
be taken as gospel.  As you can see from the discussions on
sip-implementors, this gets interesting when different devices are
traversing a proxy using different URI schemes...

  However, I suspect this won't become an issue until most SIP
implementations support SCTP.  That should be exciting! ;)


-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From faisalmaqsoodi at yahoo.com  Mon Dec  1 21:30:53 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Mon, 1 Dec 2008 21:30:53 -0800 (PST)
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <698401620812011255g2e15991w4970559ac912f9b3@mail.gmail.com>
Message-ID: <625821.37656.qm@web30706.mail.mud.yahoo.com>

Yes its not working on my system. When i copy this in default.xml dialplan, it works but as a seperate extension in dialplan/extensions it does'nt.
???????????????????????????????????????????????????????????????????????????????????????? Faisal

--- On Mon, 12/1/08, Jan Kubr <jan.kubr at gmail.com> wrote:
From: Jan Kubr <jan.kubr at gmail.com>
Subject: Re: [Freeswitch-users] How to specify Path for sound files
To: freeswitch-users at lists.freeswitch.org
Date: Monday, December 1, 2008, 12:55 PM

Hi Faisal,
the path is either an absolute path or a path relative to the
directory in the sound_prefix var in vars.xml.
So this

<action application="playback"
data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"/>

works fine on my box. You sure this one doesn't work for you?

Jan

> I tried to play a sound file using the dialplan given on the link
> http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav
>
> <action application="playback"
data="/path/to/your.wav"/>
> In place of /path/to/your.wave I used
> "/en/us/callie/misc/8000/call_secured.wav"
>
"/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
> "/sounds/en/us/callie/misc/8000/call_secured.wav"
> But none of these is useful bcoz when i call on 2009, which is
> to b dialed to play the sound, same msg is
> displayed "404 NOT FOUND"
> Plz help me out.??? Faisal

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From hads at nice.net.nz  Mon Dec  1 21:54:01 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Tue, 2 Dec 2008 18:54:01 +1300
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <625821.37656.qm@web30706.mail.mud.yahoo.com>
References: <625821.37656.qm@web30706.mail.mud.yahoo.com>
Message-ID: <200812021854.01959.hads@nice.net.nz>

On Tuesday 02 December 2008 18:30:53 Faisal Maqsoodi wrote:
> Yes its not working on my system. When i copy this in default.xml dialplan,
> it works but as a seperate extension in dialplan/extensions it does'nt.
> Faisal

It's a little hard to understand what you're saying but I'd hazard a guess 
that your extension is below the transfer to enum.

Are you creating a separate file in conf/dialplan/default/ ? What are you 
naming the file? Does it show up below the enum file in a directory listing?

As Anthony said if you set debug logging then you will see what is going on.

hads
-- 
http://nicegear.co.nz
VoIP, DVB and other Linux compatible hardware.


From dave at 3c.co.uk  Mon Dec  1 22:11:35 2008
From: dave at 3c.co.uk (David Knell)
Date: Tue, 02 Dec 2008 06:11:35 +0000
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
Message-ID: <4934D197.8080007@3c.co.uk>

Hi Anthony,
> mod_openmrcp was a contribution to the community by a 3rd party 
> individual.
>
> As i have clearly stated in 2 previous emails, the man has decided to 
> discontinue the openmrcp project.
> So now we are left with the remains of the module and discontinued 
> code.  This was not our decision it was his.
I absolutely understand this but it's important, from a user point of 
view, to be able to know which bits of FS are current/supported and 
which aren't.
> Some people use it without issue which may mean that the crash you 
> reported is windows specific and I do not have a working lab of any 
> mrcp capbable system to try it against in unix for that matter.  I 
> have a list of work to do from here to the moon and back so on an 
> issue like this, unless someone can hand me login credentials to some 
> box and give me a phone number to dial to reporduce the issue, it will 
> be a long time until we can deal with it.  
It's useful to know that there are people using mod_openmrcp without 
issue: I did ask here if anyone was a while back, and no-one fessed up.  
I'll give it a go on a Linux box and report back.  And if you'd like a 
dev/test environment set up, then just tell me which one.
> And the question arises, should we bother working on it anymore if the 
> lib has been abandoned and we cannot even get any support from it's 
> author which is where the problem most likely lies.
>
> I try not to get too annoyed by these remarks about what we *ought to 
> do* because I know people lose sight of how much of the work to 
> support the project is done by a small group of 3 people and not the 
> 2000 people it appears to be from the outside looking in. (I've been 
> answering email for 4 hours now)
Those guys who claim to have all that money in an offshore bank account 
are lying - you don't have to reply to them in future ;-)  Seriously, 
though, I don't think it's too outrageous an idea to document what's 
supported and were you (for example) to have suggested that I get in 
touch with the contributors to the various modules, ask them what their 
view of its status is, condense the answers in to a list and report 
back, it's something I'd quite happily do.
> My suggestion is to pool some cash and pay the guy to make mod_unimrcp 
> for FS that we can maintain in tree knowing the development can be 
> supported by the original author.
Quite happy to participate in that, too.. the problem is that I've a 
demo to do like yesterday and the timescale for mod_unimrcp is a bit on 
the long side for that.  I'd rather not have to do it with Asterisk and 
Lumenvox..!

Cheers --

Dave
>
>
> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk 
> <mailto:dave at 3c.co.uk>> wrote:
>
>     Hi Mike,
>
>     My experience is that it's somewhat broken - it took two trivial
>     tweaks to get it to work with IBM's ASR and TTS, but there's a
>     more intractable problem to do with memory getting overwritten (I
>     assume that this is something to do with something being freed
>     when it shouldn't be) which causes a segfault on the second or
>     third session after the module being loaded.
>
>     Without wishing to sound like a stuck record, one thing that you
>     guys really ought to do is to decide what's supported and what
>     isn't, and make this obvious - for example, move unsupported
>     modules to a different place in the tree, don't have them built by
>     default, etc.  MRCP is in the specsheet on the Wiki.  Otherwise
>     folk like Mark and I spend time installing stuff, go round in
>     circles a bit trying to make it work, and then find out (a) that
>     it doesn't and (b) it's not going to be fixed because it's not
>     supported.
>
>     Cheers --
>
>     Dave
>>     I would not say it is totally broken, it is known to work in
>>     quite a few places, but we are unlikely to be doing any new fixes
>>     in it.
>>
>>     Mike
>>
>>     On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com
>>     <mailto:mszlazak at aol.com> wrote:
>>
>>>     Hi Anthony,
>>>
>>>     Oh! OK.
>>>
>>>     So is this module "totally broken".
>>>
>>>     I say this because I can't seem to get it to work at all with
>>>     the example in that Mod_openmrcp wiki page but I thought it
>>>     might because I'm not be using the right Cepstral software
>>>     (freetrial download versus the paided for SDK) or that I'm not
>>>     using the right port numbers or something else I didn't do. I
>>>     used TcpView to look at local port associated with my Cepstral
>>>     software and changed a few things but still nothing. I changed
>>>     the loglevel setting to 7 in the wiki's example but I don't see
>>>     the kind of output on the console that I would expect for debug
>>>     mode.
>>>
>>>     Thanks. Mark.
>>>
>>>
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     David Knell, Director, 3C Limited
>     T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>     http://www.3c.co.uk 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From yudha2008 at gmail.com  Mon Dec  1 22:24:06 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 2 Dec 2008 11:54:06 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
Message-ID: <d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>

*Hi Giovanni Maruzzelli*,

To list the available devices i have given this command *pa devlist*
*output:*
freeswitch at hp30094686650.optimus.co.in> pa devlist
2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process()
Unknown Command: pa

But when i check in my system *hwconf *there is auido drives

*class: AUDIO
bus: PCI
detached: 0
driver: snd-intel8x0
desc: "Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller"
vendorId: 8086
deviceId: 24d5
subVendorId: 8086
subDeviceId: 0c4a
pciType: 1
pcidom:    0
pcibus:  0
pcidev: 1f
pcifn:  5*

How to resolve the problem. Can u correct me where i am wrong.Can u just
describe what is the error also.

Thanks for reply
*--
Warm Regards,
N.Baskar
*
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From mszlazak at aol.com  Mon Dec  1 22:40:36 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Tue, 02 Dec 2008 01:40:36 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <8CB222E46498CA1-11E0-E8B@MBLK-M05.sysops.aol.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com><191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com><8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com><CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com><4934323B.9000305@3c.co.uk><191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<8CB222E46498CA1-11E0-E8B@MBLK-M05.sysops.aol.com>
Message-ID: <8CB226004412A8C-430-1FDA@WEBMAIL-MA12.sysops.aol.com>


 


 Just to follow up.

Moshe Yudkowsky has an article on "Routing calls from FreeSwitch to Prophecy":? http://www.prophecy2006.com/node/145

My problem is that Freeswitch and Prophecy need to be on the same machine BUT both need to bind to port 5060 so I'm getting errors from one or the other depending who's running first.

So can I change what port(s) FS uses and that way avoid this conflict? 
Maybe, this might let me bridge the call via FreeSwitch to Prophecy similar to what Moshe's article discusses???


 

-----Original Message-----
From: mszlazak at aol.com
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 1 Dec 2008 4:44 pm
Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP











 





 Does "bridging" a call from FS to Voxeo's Prophecy server require openMRCP? If not then the other issue I might have is a database look up that is part of the dialogue that maybe need as the person response to prompts from the asr. It's possible to run a php script for the database stuff that Prophecy might need or could that happen via Javascript in FS? Then after the dialogue has completed I go from Prophecy back to FS.





 



-----Original Message-----

From: Anthony Minessale <anthony.minessale at gmail.com>

To: freeswitch-users at lists.freeswitch.org

Sent: Mon, 1 Dec 2008 11:17 am

Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP












mod_openmrcp was a contribution to the community by a 3rd party individual.



As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project.

So now we are left with the remains of the module and discontinued code.? This was not our decision it was his. 




Since the author of openmrcp has stated that he has a new unimrcp we are certainly going to 

work towards getting mod_unimrcp to replace mod_openmrcp.? He had already commented on that previous thread to state he is willing to consider making a new module.





Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter.? I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it.? And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies.




I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) 




My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author.







On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:






  




Hi Mike,





My experience is that it's somewhat broken - it took two trivial tweaks
to get it to work with IBM's ASR and TTS, but there's a more
intractable problem to do with memory getting overwritten (I assume
that this is something to do with something being freed when it
shouldn't be) which causes a segfault on the second or third session
after the module being loaded.





Without wishing to sound like a stuck record, one thing that you guys
really ought to do is to decide what's supported and what isn't, and
make this obvious - for example, move unsupported modules to a
different place in the tree, don't have them built by default, etc.?
MRCP is in the specsheet on the Wiki.? Otherwise folk like Mark and I
spend time installing stuff, go round in circles a bit trying to make
it work, and then find out (a) that it doesn't and (b) it's not going
to be fixed because it's not supported.





Cheers --





Dave











I would not say it is totally broken, it is known to work
in quite a few places, but we are unlikely to be doing any new fixes in
it.
  




  


  

Mike


  




  


  

On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:


  


  

    

 Hi Anthony,


    


Oh! OK.


    


So is this module "totally broken". 


    


I say this because I can't seem to get it to work at all with the
example in that Mod_openmrcp wiki page but I thought it might because
I'm not be using the right Cepstral software (freetrial download versus
the paided for SDK) or that I'm not using the right port numbers or
something else I didn't do. I used TcpView to look at local port
associated with my Cepstral software and changed a few things but still
nothing. I changed the loglevel setting to 7 in the wiki's example but
I don't see the kind of output on the console that I would expect for
debug mode.


    


Thanks. Mark.


    


    

 


    


    




    


  

  


  


  










_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org

  












-- 

David Knell, Director, 3C Limited

T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031

http://www.3c.co.uk 







_______________________________________________


Freeswitch-users mailing list


Freeswitch-users at lists.freeswitch.org


http://lists.freeswitch.org/mailman/listinfo/freeswitch-users


UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users


http://www.freeswitch.org












-- 

Anthony Minessale II



FreeSWITCH http://www.freeswitch.org/

ClueCon http://www.cluecon.com/




AIM: anthm

MSN:anthony_minessale at hotmail.com

GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com


IRC: irc.freenode.net #freeswitch



FreeSWITCH Developer Conference

sip:888 at conference.freeswitch.org

iax:guest at conference.freeswitch.org/888


googletalk:conf+888 at conference.freeswitch.org

pstn:213-799-1400





 






_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





 





 Tis the season to save your money!  Get the new AOL Holiday Toolbar for money saving offers and gift ideas. 



 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From faisalmaqsoodi at yahoo.com  Mon Dec  1 22:44:21 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Mon, 1 Dec 2008 22:44:21 -0800 (PST)
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <200812021854.01959.hads@nice.net.nz>
Message-ID: <603330.68689.qm@web30704.mail.mud.yahoo.com>

Actually i copied the following text in a new text file and saved it as test1.xml file in /conf/dialplan/extensions, where 99999_enum.xml and 00_pizza_demo.xml exist, but it didnt worked.
<extension name="wavs">
    <condition field="destination_number" expression="^2009$">
      <action application="playback" data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"/>
    </condition>
  </extension>
Then i copied the same text and pasted in conf/dialplan/default.xml file below the line
<context name="default"> and above the line
<extension name="unloop">
and it worked successfully. Hope i ve explained what i wanted to.
??????????????????????????????????????????????????????????????????????????????????????? Faisal

--- On Mon, 12/1/08, Hadley Rich <hads at nice.net.nz> wrote:
From: Hadley Rich <hads at nice.net.nz>
Subject: Re: [Freeswitch-users] How to specify Path for sound files
To: freeswitch-users at lists.freeswitch.org
Date: Monday, December 1, 2008, 9:54 PM

On Tuesday 02 December 2008 18:30:53 Faisal Maqsoodi wrote:
> Yes its not working on my system. When i copy this in default.xml
dialplan,
> it works but as a seperate extension in dialplan/extensions it
does'nt.
> Faisal

It's a little hard to understand what you're saying but I'd hazard
a guess 
that your extension is below the transfer to enum.

Are you creating a separate file in conf/dialplan/default/ ? What are you 
naming the file? Does it show up below the enum file in a directory listing?

As Anthony said if you set debug logging then you will see what is going on.

hads
-- 
http://nicegear.co.nz
VoIP, DVB and other Linux compatible hardware.

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From gmaruzz at celliax.org  Mon Dec  1 22:50:56 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 2 Dec 2008 07:50:56 +0100
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
Message-ID: <7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>

Hi Baskar,

you have to compile and enable the module mod_portaudio.

Please edit the modules.conf in the main directory of the FS sources,
and remove the "#" before mod_portaudio. Also, after compilation and
installation ("make install"), in the directory
/usr/local/freeswitch/conf/autoload/ edit the file modules.conf.xml so
to enable the portaudio module and edit the portaudio.conf.xml to
reflect your setup.

Sincerely,

Giovanni Maruzzelli

=========================================
Cell : 39-347-2665618
Fax : 39-02-87390039



On Tue, Dec 2, 2008 at 7:24 AM, Baskar <yudha2008 at gmail.com> wrote:
> Hi Giovanni Maruzzelli,
>
> To list the available devices i have given this command pa devlist
> output:
> freeswitch at hp30094686650.optimus.co.in> pa devlist
> 2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process()
> Unknown Command: pa
>
> But when i check in my system hwconf there is auido drives
>
> class: AUDIO
> bus: PCI
> detached: 0
> driver: snd-intel8x0
> desc: "Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller"
> vendorId: 8086
> deviceId: 24d5
> subVendorId: 8086
> subDeviceId: 0c4a
> pciType: 1
> pcidom:    0
> pcibus:  0
> pcidev: 1f
> pcifn:  5
>
> How to resolve the problem. Can u correct me where i am wrong.Can u just
> describe what is the error also.
>
> Thanks for reply
> --
> Warm Regards,
> N.Baskar
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mszlazak at aol.com  Mon Dec  1 23:00:32 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Tue, 02 Dec 2008 02:00:32 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
Message-ID: <8CB2262CD41B5B4-430-201D@WEBMAIL-MA12.sysops.aol.com>



 I need to barge in again and add to my last post with this email from Voxeo support. Here is their response to the port binding conflict and it brings up a possible problem if FreeSwitch will be looking for Prophecy at that port? I assumed it would if I set up the extension right but now I don't know and need your assistance with this issue ... as well. 

Thank you.

MESSAGE:
Hi Mark,

You are correct in that having multiple applications binding to the same port 
can cause a bundle of problems. You can configure Prophecy to stay away from 
port 5060, but then the question is whether FreeSwitch will be looking for 
Prophecy at that port (if its assuming that it's residing on a different box). 
Port 5060 is the standard for SIP traffic.

To get Prophecy off 5060 you will need to edit the config.xml and 
callrouting.xml files. You will need to search out all instances of "5060" and 
replace with, perhaps, port 5068. For instance:

<item name="ListenOnIP1">0.0.0.0:5068</item>
<item name="ListenOnIP2">0.0.0.0:5061</item>
<item name="ListenOnIP3">0.0.0.0:5067</item>
<item name="ListenOnIP4">0.0.0.0:5063</item>
<item name="ListenOnIP5">0.0.0.0:5064</item>
<item name="ListenOnIP6">0.0.0.0:5065</item>

instead of this...

<item name="ListenOnIP1">0.0.0.0:5060</item>
<item name="ListenOnIP2">0.0.0.0:5061</item>
<item name="ListenOnIP3">0.0.0.0:5062</item>
<item name="ListenOnIP4">0.0.0.0:5063</item>
<item name="ListenOnIP5">0.0.0.0:5064</item>
<item name="ListenOnIP6">0.0.0.0:5065</item>

Regards,
Jeff Kustermann
Voxeo Support




? 





 





 



-----Original Message-----

From: mszlazak at aol.com

To: freeswitch-users at lists.freeswitch.org

Sent: Mon, 1 Dec 2008 10:40 pm

Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP















 








 Just to follow up.





Moshe Yudkowsky has an article on "Routing calls from FreeSwitch to Prophecy":? http://www.prophecy2006.com/node/145





My problem is that Freeswitch and Prophecy need to be on the same machine BUT both need to bind to port 5060 so I'm getting errors from one or the other depending who's running first.





So can I change what port(s) FS uses and that way avoid this conflict? 


Maybe, this might let me bridge the call via FreeSwitch to Prophecy similar to what Moshe's article discusses???








 





-----Original Message-----


From: mszlazak at aol.com


To: freeswitch-users at lists.freeswitch.org


Sent: Mon, 1 Dec 2008 4:44 pm


Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP



















 











 Does "bridging" a call from FS to Voxeo's Prophecy server require openMRCP? If not then the other issue I might have is a database look up that is part of the dialogue that maybe need as the person response to prompts from the asr. It's possible to run a php script for the database stuff that Prophecy might need or could that happen via Javascript in FS? Then after the dialogue has completed I go from Prophecy back to FS.











 







-----Original Message-----



From: Anthony Minessale <anthony.minessale at gmail.com>



To: freeswitch-users at lists.freeswitch.org



Sent: Mon, 1 Dec 2008 11:17 am



Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP


















mod_openmrcp was a contribution to the community by a 3rd party individual.







As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project.



So now we are left with the remains of the module and discontinued code.? This was not our decision it was his. 








Since the author of openmrcp has stated that he has a new unimrcp we are certainly going to 



work towards getting mod_unimrcp to replace mod_openmrcp.? He had already commented on that previous thread to state he is willing to consider making a new module.









Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter.? I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it.? And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies.








I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) 








My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author.















On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:








  






Hi Mike,









My experience is that it's somewhat broken - it took two trivial tweaks
to get it to work with IBM's ASR and TTS, but there's a more
intractable problem to do with memory getting overwritten (I assume
that this is something to do with something being freed when it
shouldn't be) which causes a segfault on the second or third session
after the module being loaded.









Without wishing to sound like a stuck record, one thing that you guys
really ought to do is to decide what's supported and what isn't, and
make this obvious - for example, move unsupported modules to a
different place in the tree, don't have them built by default, etc.?
MRCP is in the specsheet on the Wiki.? Otherwise folk like Mark and I
spend time installing stuff, go round in circles a bit trying to make
it work, and then find out (a) that it doesn't and (b) it's not going
to be fixed because it's not supported.









Cheers --









Dave





















I would not say it is totally broken, it is known to work
in quite a few places, but we are unlikely to be doing any new fixes in
it.
  








  




  



Mike




  








  




  



On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:




  




  

    



 Hi Anthony,




    




Oh! OK.




    




So is this module "totally broken". 




    




I say this because I can't seem to get it to work at all with the
example in that Mod_openmrcp wiki page but I thought it might because
I'm not be using the right Cepstral software (freetrial download versus
the paided for SDK) or that I'm not using the right port numbers or
something else I didn't do. I used TcpView to look at local port
associated with my Cepstral software and changed a few things but still
nothing. I changed the loglevel setting to 7 in the wiki's example but
I don't see the kind of output on the console that I would expect for
debug mode.




    




Thanks. Mark.




    




    



 




    




    








    




  

  




  




  


























_______________________________________________







Freeswitch-users mailing list







Freeswitch-users at lists.freeswitch.org







http://lists.freeswitch.org/mailman/listinfo/freeswitch-users







UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users







http://www.freeswitch.org







  






















-- 







David Knell, Director, 3C Limited







T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031







http://www.3c.co.uk 











_______________________________________________




Freeswitch-users mailing list




Freeswitch-users at lists.freeswitch.org




http://lists.freeswitch.org/mailman/listinfo/freeswitch-users




UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users




http://www.freeswitch.org






















-- 



Anthony Minessale II







FreeSWITCH http://www.freeswitch.org/



ClueCon http://www.cluecon.com/








AIM: anthm



MSN:anthony_minessale at hotmail.com



GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com




IRC: irc.freenode.net #freeswitch







FreeSWITCH Developer Conference



sip:888 at conference.freeswitch.org



iax:guest at conference.freeswitch.org/888




googletalk:conf+888 at conference.freeswitch.org



pstn:213-799-1400









 








_______________________________________________







Freeswitch-users mailing list







Freeswitch-users at lists.freeswitch.org







http://lists.freeswitch.org/mailman/listinfo/freeswitch-users







UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users







http://www.freeswitch.org













 









 Tis the season to save your money!  Get the new AOL Holiday Toolbar for money saving offers and gift ideas. 







 







_______________________________________________



Freeswitch-users mailing list



Freeswitch-users at lists.freeswitch.org



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



http://www.freeswitch.org








 







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_______________________________________________

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Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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From hads at nice.net.nz  Mon Dec  1 23:03:35 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Tue, 2 Dec 2008 20:03:35 +1300
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <603330.68689.qm@web30704.mail.mud.yahoo.com>
References: <603330.68689.qm@web30704.mail.mud.yahoo.com>
Message-ID: <200812022003.36345.hads@nice.net.nz>

On Tuesday 02 December 2008 19:44:21 Faisal Maqsoodi wrote:
> Actually i copied the following text in a new text file and saved it as
> test1.xml file in /conf/dialplan/extensions, where 99999_enum.xml and
> 00_pizza_demo.xml exist, but it didnt worked. <extension name="wavs">
>     <condition field="destination_number" expression="^2009$">
>       <action application="playback"
> data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
>/> </condition>
>   </extension>
> Then i copied the same text and pasted in conf/dialplan/default.xml file
> below the line <context name="default"> and above the line
> <extension name="unloop">
> and it worked successfully. Hope i ve explained what i wanted to.
> ???????????????????????????????????????????????????????????????????????????
>???????????? Faisal
>
> --- On Mon, 12/1/08, Hadley Rich <hads at nice.net.nz> wrote:
> From: Hadley Rich <hads at nice.net.nz>
> Subject: Re: [Freeswitch-users] How to specify Path for sound files
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, December 1, 2008, 9:54 PM
>
> On Tuesday 02 December 2008 18:30:53 Faisal Maqsoodi wrote:
> > Yes its not working on my system. When i copy this in default.xml
>
> dialplan,
>
> > it works but as a seperate extension in dialplan/extensions it
>
> does'nt.
>
> > Faisal
>
> It's a little hard to understand what you're saying but I'd hazard
> a guess
> that your extension is below the transfer to enum.
>
> Are you creating a separate file in conf/dialplan/default/ ? What are you
> naming the file? Does it show up below the enum file in a directory
> listing?
>
> As Anthony said if you set debug logging then you will see what is going
> on.
>
> hads

People should really quote properly huh.

As I suspected you're getting transferred to enum (which the debug logging 
would have told you).

Try naming your file 50-test.xml and seeing what happens.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier


From faisalmaqsoodi at yahoo.com  Mon Dec  1 23:13:41 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Mon, 1 Dec 2008 23:13:41 -0800 (PST)
Subject: [Freeswitch-users] How to specify Path for sound files [DONE]
In-Reply-To: <200812022003.36345.hads@nice.net.nz>
Message-ID: <9442.5309.qm@web30706.mail.mud.yahoo.com>

Sir thank you very much. It really works.
?????????????????????????????????????????????????????????? Faisal

--- On Mon, 12/1/08, Hadley Rich <hads at nice.net.nz> wrote:
From: Hadley Rich <hads at nice.net.nz>
Subject: Re: [Freeswitch-users] How to specify Path for sound files
To: freeswitch-users at lists.freeswitch.org
Date: Monday, December 1, 2008, 11:03 PM

On Tuesday 02 December 2008 19:44:21 Faisal Maqsoodi wrote:
> Actually i copied the following text in a new text file and saved it as
> test1.xml file in /conf/dialplan/extensions, where 99999_enum.xml and
> 00_pizza_demo.xml exist, but it didnt worked. <extension
name="wavs">
>     <condition field="destination_number"
expression="^2009$">
>       <action application="playback"
>
data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
>/> </condition>
>   </extension>
> Then i copied the same text and pasted in conf/dialplan/default.xml file
> below the line <context name="default"> and above the line
> <extension name="unloop">
> and it worked successfully. Hope i ve explained what i wanted to.
>
???????????????????????????????????????????????????????????????????????????
>???????????? Faisal
>
> --- On Mon, 12/1/08, Hadley Rich <hads at nice.net.nz> wrote:
> From: Hadley Rich <hads at nice.net.nz>
> Subject: Re: [Freeswitch-users] How to specify Path for sound files
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, December 1, 2008, 9:54 PM
>
> On Tuesday 02 December 2008 18:30:53 Faisal Maqsoodi wrote:
> > Yes its not working on my system. When i copy this in default.xml
>
> dialplan,
>
> > it works but as a seperate extension in dialplan/extensions it
>
> does'nt.
>
> > Faisal
>
> It's a little hard to understand what you're saying but I'd
hazard
> a guess
> that your extension is below the transfer to enum.
>
> Are you creating a separate file in conf/dialplan/default/ ? What are you
> naming the file? Does it show up below the enum file in a directory
> listing?
>
> As Anthony said if you set debug logging then you will see what is going
> on.
>
> hads

People should really quote properly huh.

As I suspected you're getting transferred to enum (which the debug logging 
would have told you).

Try naming your file 50-test.xml and seeing what happens.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From yudha2008 at gmail.com  Mon Dec  1 23:27:15 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 2 Dec 2008 12:57:15 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
Message-ID: <d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>

*Hi,

I have updated all the above events you told .It's working fine but when i
call extension 1002 from freeswitch console, call is connected to extension
1002, but FS is aborted but call is established in1002. what shall i do.
what was the error.

Full freeswitch get cut.*

*output:*
freeswitch at hp30094686650.optimus.co.in>* pa call 1002*
2008-12-02 12:54:05 [NOTICE] switch_channel.c:553 switch_channel_set_name()
New Channel portaudio/1002 [20b1163a-29c7-4369-bdb5-27398dc1a263]
2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call() Channel
[portaudio/1002] has been answered
API CALL [pa(call 1002)] output:
SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263

2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing
FreeSWITCH->1002 in context default
2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805 switch_ivr_set_user() can't
find user [default@]
freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]
mod_dptools.c:902 info_function() CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [portaudio/1002]
Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [L16]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [L16]
Channel-Write-Codec-Rate: [8000]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [FreeSWITCH]
Caller-Caller-ID-Number: [0000000000]
Caller-Network-Addr: [172.20.176.32]
Caller-Destination-Number: [1002]
Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
Caller-Source: [mod_portaudio]
Caller-Context: [default]
Caller-Channel-Name: [portaudio/1002]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1228202645898038]
Caller-Channel-Created-Time: [1228202645898038]
Caller-Channel-Answered-Time: [1228202647630133]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_channel_name: [portaudio/1002]
variable_endpoint_disposition: [ANSWER]
variable_read_codec: [L16]
variable_read_rate: [8000]
variable_write_codec: [L16]
variable_write_rate: [8000]
variable_use_profile: [nat]
variable_dialed_ext: [1002]
variable_current_application: [info]


2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer-State []n
2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML
features
2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2
record_session::/usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML
features
2008-12-02 12:54:07 [NOTICE] switch_channel.c:553 switch_channel_set_name()
New Channel sofia/internal/1002 at 172.20.179.201:23878;rinstance=de482996ac747c8d
[f7f80a05-be75-414b-bcea-4e5a34c3351e]
*freeswitch: src/switch_core_io.c:179: switch_core_session_read_frame:
Assertion `(*frame)->codec != ((void *)0)' failed.
Aborted (core dumped)
[root at hp30094686650 bin]#
*
*Thanks for the reply. Correct me were i am wrong.*

*Warm Regards,
N.Baskar
*
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From msc at freeswitch.org  Mon Dec  1 23:35:58 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 1 Dec 2008 23:35:58 -0800
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
Message-ID: <72441BC8-74C6-4490-B025-278E8C3F0CCF@freeswitch.org>

Does the core dump always happen in this call scenario? If so, can you  
get a back trace? Put it on pastebin. That will hopefully help narrow  
down the issue.

-MC

Sent from my iPhone

On Dec 1, 2008, at 11:27 PM, Baskar <yudha2008 at gmail.com> wrote:

> Hi,
>
> I have updated all the above events you told .It's working fine but  
> when i call extension 1002 from freeswitch console, call is  
> connected to extension 1002, but FS is aborted but call is  
> established in1002. what shall i do. what was the error.
>
> Full freeswitch get cut.
>
> output:
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-02 12:54:05 [NOTICE] switch_channel.c:553  
> switch_channel_set_name() New Channel portaudio/1002  
> [20b1163a-29c7-4369-bdb5-27398dc1a263]
> 2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call()  
> Channel [portaudio/1002] has been answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263
>
> 2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt()  
> Processing FreeSWITCH->1002 in context default
> 2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805  
> switch_ivr_set_user() can't find user [default@]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]  
> mod_dptools.c:902 info_function() CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228202645898038]
> Caller-Channel-Created-Time: [1228202645898038]
> Caller-Channel-Answered-Time: [1228202647630133]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer- 
> State []n
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1  
> execute_extension::dx XML features
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ 
> usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3  
> execute_extension::cf XML features
> 2008-12-02 12:54:07 [NOTICE] switch_channel.c:553  
> switch_channel_set_name() New Channel sofia/internal/1002 at 172.20.179.201 
> :23878;rinstance=de482996ac747c8d [f7f80a05-be75-414b- 
> bcea-4e5a34c3351e]
> freeswitch: src/switch_core_io.c:179:  
> switch_core_session_read_frame: Assertion `(*frame)->codec != ((void  
> *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
> Thanks for the reply. Correct me were i am wrong.
>
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gmaruzz at celliax.org  Mon Dec  1 23:39:38 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 2 Dec 2008 08:39:38 +0100
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
Message-ID: <7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>

Baskar,

that is bizarre.

Seems there is a problem with mod_sofia, the module that manages SIP
connection to the SIP client at 1002 extension.

Maybe someone else on the list can be of more help.

Sincerely,

Giovanni Maruzzelli

=========================================
Cell : 39-347-2665618
Fax : 39-02-87390039



On Tue, Dec 2, 2008 at 8:27 AM, Baskar <yudha2008 at gmail.com> wrote:
> Hi,
>
> I have updated all the above events you told .It's working fine but when i
> call extension 1002 from freeswitch console, call is connected to extension
> 1002, but FS is aborted but call is established in1002. what shall i do.
> what was the error.
>
> Full freeswitch get cut.
>
> output:
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-02 12:54:05 [NOTICE] switch_channel.c:553 switch_channel_set_name()
> New Channel portaudio/1002 [20b1163a-29c7-4369-bdb5-27398dc1a263]
> 2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call() Channel
> [portaudio/1002] has been answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263
>
> 2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing
> FreeSWITCH->1002 in context default
> 2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805 switch_ivr_set_user() can't
> find user [default@]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]
> mod_dptools.c:902 info_function() CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228202645898038]
> Caller-Channel-Created-Time: [1228202645898038]
> Caller-Channel-Answered-Time: [1228202647630133]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer-State []n
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML
> features
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2
> record_session::/usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML
> features
> 2008-12-02 12:54:07 [NOTICE] switch_channel.c:553 switch_channel_set_name()
> New Channel
> sofia/internal/1002 at 172.20.179.201:23878;rinstance=de482996ac747c8d
> [f7f80a05-be75-414b-bcea-4e5a34c3351e]
> freeswitch: src/switch_core_io.c:179: switch_core_session_read_frame:
> Assertion `(*frame)->codec != ((void *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
> Thanks for the reply. Correct me were i am wrong.
>
> Warm Regards,
> N.Baskar
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From r.pankratz at fh-wolfenbuettel.de  Tue Dec  2 01:12:13 2008
From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz)
Date: Tue, 02 Dec 2008 10:12:13 +0100
Subject: [Freeswitch-users] Dialing tone when placing a call with portaudio
In-Reply-To: <20081120154859.16192.qmail@server15.citromail.hu>
References: <20081120154859.16192.qmail@server15.citromail.hu>
Message-ID: <4934FBED.7030307@fh-wolfenbuettel.de>

Hello,
when using mod_portaudio for calling somebody I don't hear anything 
until the other party answers the call. Is it possible to play a dialing 
tone when the other party is ringing?

Best regards
   Ren? Pankratz



From mike at jerris.com  Tue Dec  2 02:49:40 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 05:49:40 -0500
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
Message-ID: <3F8D37CC-261F-4C5E-A9F3-64F9AD97F761@jerris.com>

What revision of freeswitch is this?  Can you please test this with  
svn trunk?

Mike

On Dec 2, 2008, at 2:27 AM, Baskar wrote:

> Hi,
>
> I have updated all the above events you told .It's working fine but  
> when i call extension 1002 from freeswitch console, call is  
> connected to extension 1002, but FS is aborted but call is  
> established in1002. what shall i do. what was the error.
>
> Full freeswitch get cut.
>
> output:
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-02 12:54:05 [NOTICE] switch_channel.c:553  
> switch_channel_set_name() New Channel portaudio/1002  
> [20b1163a-29c7-4369-bdb5-27398dc1a263]
> 2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call()  
> Channel [portaudio/1002] has been answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263
>
> 2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt()  
> Processing FreeSWITCH->1002 in context default
> 2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805  
> switch_ivr_set_user() can't find user [default@]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]  
> mod_dptools.c:902 info_function() CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228202645898038]
> Caller-Channel-Created-Time: [1228202645898038]
> Caller-Channel-Answered-Time: [1228202647630133]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer- 
> State []n
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1  
> execute_extension::dx XML features
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ 
> usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3  
> execute_extension::cf XML features
> 2008-12-02 12:54:07 [NOTICE] switch_channel.c:553  
> switch_channel_set_name() New Channel sofia/internal/1002 at 172.20.179.201 
> :23878;rinstance=de482996ac747c8d [f7f80a05-be75-414b- 
> bcea-4e5a34c3351e]
> freeswitch: src/switch_core_io.c:179:  
> switch_core_session_read_frame: Assertion `(*frame)->codec != ((void  
> *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
> Thanks for the reply. Correct me were i am wrong.
>
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Dec  2 02:51:59 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 05:51:59 -0500
Subject: [Freeswitch-users] Dialing tone when placing a call with
	portaudio
In-Reply-To: <4934FBED.7030307@fh-wolfenbuettel.de>
References: <20081120154859.16192.qmail@server15.citromail.hu>
	<4934FBED.7030307@fh-wolfenbuettel.de>
Message-ID: <ADE59C30-1DBE-4DB5-B96B-9EDA0A0DCE07@jerris.com>

What are you calling, sip I assume, this may be a case where the sip  
signaling is sending a 180 ringing instead of a 183 and we are not  
generating ringback in that case.  Can you please confirm that and  
test if setting the ringback channel variable before bridge fixes this  
issue?

Mike

On Dec 2, 2008, at 4:12 AM, Rene Pankratz wrote:

> Hello,
> when using mod_portaudio for calling somebody I don't hear anything
> until the other party answers the call. Is it possible to play a  
> dialing
> tone when the other party is ringing?



From woodydickson at gmail.com  Tue Dec  2 02:55:06 2008
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 2 Dec 2008 18:55:06 +0800
Subject: [Freeswitch-users] libfreeswitch question
Message-ID: <a44688a0812020255n2e1c7407ydb5145e025ce34e6@mail.gmail.com>

Hi,

I am just having a dumb question and hoping someone can help me.  I am
trying to run a c program with libfreeswitch embedded so I can use some
external mechanism to keep track of freeswitch, but I am having problem
while compiling:

[root at localhost fs]# gcc switchnode.c -I/usr/local/freeswitch/include
-L/usr/local/freeswitch/lib -lfreeswitch -lpthread
switchnode.c: In function 'main':
switchnode.c:11: warning: passing argument 1 of
'switch_core_init_and_modload' makes integer from pointer without a cast
switchnode.c:11: warning: passing argument 3 of
'switch_core_init_and_modload' from incompatible pointer type
/usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to
`clock_gettime'
/usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to
`uuid_generate'
/usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `crypt_r'
collect2: ld returned 1 exit status
[root at localhost fs]#


Does anyone know which library is missing?

Thanks,
Woody
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From yudha2008 at gmail.com  Tue Dec  2 02:57:27 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 2 Dec 2008 16:27:27 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
Message-ID: <d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>

Hi,

*After starting the freeswitch I try to dial a extension from console* *but
when i call extension 1002 from freeswitch console, call is connected to
extension 1002, but FS is aborted but call is established in1002. What shall
I do? What was the error?*
*
 I have pasted the console events in pastebin in this path:

**http://fr.pastebin.ca/1273382
**
What is the error? Can any one correct me where I am wrong and try to
resolve the problem.

I want to know why Fs Aborted what should be done to recover from Aborted.*


-- 
Warm Regards,
N.Baskar
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From mike at jerris.com  Tue Dec  2 03:00:25 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 06:00:25 -0500
Subject: [Freeswitch-users] libfreeswitch question
In-Reply-To: <a44688a0812020255n2e1c7407ydb5145e025ce34e6@mail.gmail.com>
References: <a44688a0812020255n2e1c7407ydb5145e025ce34e6@mail.gmail.com>
Message-ID: <0E13D02B-53C1-453C-A09C-5002749A9BB4@jerris.com>


On Dec 2, 2008, at 5:55 AM, Woody Dickson wrote:

> Hi,
>
> I am just having a dumb question and hoping someone can help me.  I  
> am trying to run a c program with libfreeswitch embedded so I can  
> use some external mechanism to keep track of freeswitch, but I am  
> having problem while compiling:
>
> [root at localhost fs]# gcc switchnode.c -I/usr/local/freeswitch/ 
> include -L/usr/local/freeswitch/lib -lfreeswitch -lpthread
> switchnode.c: In function 'main':
> switchnode.c:11: warning: passing argument 1 of  
> 'switch_core_init_and_modload' makes integer from pointer without a  
> cast
> switchnode.c:11: warning: passing argument 3 of  
> 'switch_core_init_and_modload' from incompatible pointer type

looks like you have the wrong var types you are passing here.

>
> /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to  
> `clock_gettime'

-lrt

>
> /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to  
> `uuid_generate'

-luuid

>
> /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to  
> `crypt_r'

-lcrypt

>
> collect2: ld returned 1 exit status
> [root at localhost fs]#
>
>
> Does anyone know which library is missing?



From mike at jerris.com  Tue Dec  2 03:02:05 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 06:02:05 -0500
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
Message-ID: <59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>

This appears to be a somewhat older version of svn trunk.  Please re- 
test with current svn trunk

Thanks
Mike

On Dec 2, 2008, at 5:57 AM, Baskar wrote:

> Hi,
>
> After starting the freeswitch I try to dial a extension from console  
> but when i call extension 1002 from freeswitch console, call is  
> connected to extension 1002, but FS is aborted but call is  
> established in1002. What shall I do? What was the error?
>
>  I have pasted the console events in pastebin in this path:
>
> http://fr.pastebin.ca/1273382
>
> What is the error? Can any one correct me where I am wrong and try  
> to resolve the problem.
>
> I want to know why Fs Aborted what should be done to recover from  
> Aborted.

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From yudha2008 at gmail.com  Tue Dec  2 03:07:20 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 2 Dec 2008 16:37:20 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
Message-ID: <d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>

*Hi,

This is the svn version i have installed before a month

FreeSWITCH Version 1.0.trunk (10130M)
*
-- 
*Warm Regards,
N.Baskar*
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From keith.wood2000 at gmail.com  Tue Dec  2 03:23:14 2008
From: keith.wood2000 at gmail.com (Keith Wood)
Date: Tue, 2 Dec 2008 19:23:14 +0800
Subject: [Freeswitch-users] Problem with Freeswitch capturing DTMF
Message-ID: <b7737fe70812020323l23c93a17i2761b2947dc6b20b@mail.gmail.com>

Hi,

I am wondering if I am the only one getting this problem or not.  When
sending in DTMF to freeswitch, freeswitch is not always capable of capturing
all the DTMF being sent.  For instance, sending 1000 to freeswitch may end
up becoming 100 or 10003 becoming 1003.  Am I the only one getting this
strange issue?

If anyone know how to fix this problem, I would greatly appreciate it.

Regards,
Keith
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From mike at jerris.com  Tue Dec  2 03:45:28 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 06:45:28 -0500
Subject: [Freeswitch-users] Problem with Freeswitch capturing DTMF
In-Reply-To: <b7737fe70812020323l23c93a17i2761b2947dc6b20b@mail.gmail.com>
References: <b7737fe70812020323l23c93a17i2761b2947dc6b20b@mail.gmail.com>
Message-ID: <BBE9EE51-0FFD-400C-9AC1-4E93395D6080@jerris.com>

We generally are as good as possible on capturing dtmf reliably.  If  
you are seeing dropouts like that I would have to guess that this is a  
very lossy line.  Could you try and look at the packet capture of a  
call that is missing digits and see if you are indeed dropping a lot  
of packets.  If this is the case you could try info dtmf although that  
method has it's own issues.

Mike

On Dec 2, 2008, at 6:23 AM, "Keith Wood" <keith.wood2000 at gmail.com>  
wrote:

> Hi,
>
> I am wondering if I am the only one getting this problem or not.   
> When sending in DTMF to freeswitch, freeswitch is not always capable  
> of capturing all the DTMF being sent.  For instance, sending 1000 to  
> freeswitch may end up becoming 100 or 10003 becoming 1003.  Am I the  
> only one getting this strange issue?
>
> If anyone know how to fix this problem, I would greatly appreciate it.
>
> Regards,
> Keith
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From dave at 3c.co.uk  Tue Dec  2 04:03:52 2008
From: dave at 3c.co.uk (David Knell)
Date: Tue, 02 Dec 2008 12:03:52 +0000
Subject: [Freeswitch-users] Problem with Freeswitch capturing DTMF
In-Reply-To: <BBE9EE51-0FFD-400C-9AC1-4E93395D6080@jerris.com>
References: <b7737fe70812020323l23c93a17i2761b2947dc6b20b@mail.gmail.com>
	<BBE9EE51-0FFD-400C-9AC1-4E93395D6080@jerris.com>
Message-ID: <49352428.4040706@3c.co.uk>

Hi Keith,

I was just writing a note along similar lines to Mike's.  If you need a 
hand getting
a packet capture or interpreting it, drop me a note off-list.

Cheers --

Dave

> We generally are as good as possible on capturing dtmf reliably.  If  
> you are seeing dropouts like that I would have to guess that this is a  
> very lossy line.  Could you try and look at the packet capture of a  
> call that is missing digits and see if you are indeed dropping a lot  
> of packets.  If this is the case you could try info dtmf although that  
> method has it's own issues.
>
> Mike
>
> On Dec 2, 2008, at 6:23 AM, "Keith Wood" <keith.wood2000 at gmail.com>  
> wrote:
>
>   
>> Hi,
>>
>> I am wondering if I am the only one getting this problem or not.   
>> When sending in DTMF to freeswitch, freeswitch is not always capable  
>> of capturing all the DTMF being sent.  For instance, sending 1000 to  
>> freeswitch may end up becoming 100 or 10003 becoming 1003.  Am I the  
>> only one getting this strange issue?
>>
>> If anyone know how to fix this problem, I would greatly appreciate it.
>>
>> Regards,
>> Keith
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From woodydickson at gmail.com  Tue Dec  2 04:05:33 2008
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 2 Dec 2008 20:05:33 +0800
Subject: [Freeswitch-users] Question about wrapping libfreeswitch
Message-ID: <a44688a0812020405k387df840k1ed1a33da370ab6f@mail.gmail.com>

Hi,

I am sorry again for sending another email to the group again.  I am working
on embedding libfreeswitch to provide better monitoring.  The first thing I
attempt to do is to run the sample  code provided in the website:

 #include <switch.h>
 int main(int argc, char **argv)
 {
   switch_core_flag_t flags = SCF_USE_SQL;
   int nc=0; /* this is for 'no console' mode, FALSE console is there, TRUE
it isnt */
   const char **err = NULL;     /* error value for return from freeswitch
initialization */
   #define LOGFILE "freeswitch.log"
   static char *lfile = LOGFILE; /* if NULL no logfile is generated */

   switch_core_init_and_modload(*lfile,flags,err);
   switch_core_runtime_loop(nc);
   switch_core_destroy();

   return (0); /* per C89 spec */
 }

But this code gives me segmentation fault when executing it.  This piece of
code is supposed to start up freeswitch and run it is a loop.  Does anyone
see what is wrong with it?  Does anyone have any working example that I can
refer to?

Thanks,
Woody
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From mike at jerris.com  Tue Dec  2 04:29:13 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 07:29:13 -0500
Subject: [Freeswitch-users] Question about wrapping libfreeswitch
In-Reply-To: <a44688a0812020405k387df840k1ed1a33da370ab6f@mail.gmail.com>
References: <a44688a0812020405k387df840k1ed1a33da370ab6f@mail.gmail.com>
Message-ID: <9719A132-2855-4D8D-BBE1-C64269E54C0D@jerris.com>

I think the api changed a little bit for this.  The easiest starting  
point would be to just clone switch.c and chop out any of the stuff  
you don't need, it's mostly argument handling code in there.

Mike

On Dec 2, 2008, at 7:05 AM, "Woody Dickson" <woodydickson at gmail.com>  
wrote:

> Hi,
>
> I am sorry again for sending another email to the group again.  I am  
> working on embedding libfreeswitch to provide better monitoring.   
> The first thing I attempt to do is to run the sample  code provided  
> in the website:
>
>  #include <switch.h>
>  int main(int argc, char **argv)
>  {
>    switch_core_flag_t flags = SCF_USE_SQL;
>    int nc=0; /* this is for 'no console' mode, FALSE console is  
> there, TRUE it isnt */
>    const char **err = NULL;     /* error value for return from  
> freeswitch  initialization */
>    #define LOGFILE "freeswitch.log"
>    static char *lfile = LOGFILE; /* if NULL no logfile is generated */
>
>    switch_core_init_and_modload(*lfile,flags,err);
>    switch_core_runtime_loop(nc);
>    switch_core_destroy();
>
>    return (0); /* per C89 spec */
>  }
>
> But this code gives me segmentation fault when executing it.  This  
> piece of code is supposed to start up freeswitch and run it is a  
> loop.  Does anyone see what is wrong with it?  Does anyone have any  
> working example that I can refer to?
>
> Thanks,
> Woody
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From c_cav_01 at yahoo.com  Tue Dec  2 05:25:07 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 05:25:07 -0800 (PST)
Subject: [Freeswitch-users]  Wrong # in voicemail
Message-ID: <20791453.post@talk.nabble.com>


My dialplan is pretty simple.  I have a single trunk with a vonage softphone
DID (1303... we'll call it main) and a "virtual" DID (1816...) which rings
the softphone DID.  All incoming calls show up as from softphone DID but the
sip_to_user holds the actual number dialed so I can enter the dialplan
properly.

I have 2 extensions in my directory/extensions, one for each of the DID's. 
The extensions check sip_to_user for match and that works great.  I match on
([0,1]?)(<10 digit did>) and it enters the dialplans correctly, plays the
right music for each DID while the dial is occuring, so all that works.  The
bridge to user/$2@$${domain} also works fine.

The continue_on_fail is set properly so on no answer call_timeout hits (at
25 secs), and goes to voicemail...  works also for both numbers.

transfer to voicemail is as follows
 <action application="answer"/>
 <action application="voicemail" data="default $${domain} $2"/>

which should be pulling $2 from the condition check shown above, which it
does, cuz the bridge works...

When I call in on main DID, I get "leave a message for 1303..."  The Main
DID..
When I call in on virtual, I get "leave a message for 1303..."  The Main DID
rather than the 1816....

How can I get voicemail to use the correct DID.    HELP!!  :D
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20791453.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From odermann at googlemail.com  Tue Dec  2 05:48:15 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 14:48:15 +0100
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
Message-ID: <5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>

we configured mod_shout and are able to record mp3. but if we start to
playback the file, it will only be played back to that point, which
was recorded, when we started the player.
we do this with "api uuid_record uuid start /var/www/test.mp3".

we are also able to playback a (radio-)stream to an uuid with
shout://ip-adress:12345

but what do we have to do, to listen to the file/stream with a player?

it seems, that fs has to stream to recording file to a streaming
server (like icecast), right? but if we do "api uuid_record uuid start
shout://user:passwd at ip-adress:12345/" (and other combinations), we get
an error:
2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid URL: xxxxx
2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851
switch_ivr_record_session() Error opening shout://xxxx

are we on the right track? is there something else we have to do to
make it work?


thanks for your help.



2008/12/1 Anthony Minessale <anthony.minessale at gmail.com>:
> yes,
>
> mod_shout will broadcast calls as MP3 that you can listen to in
> itunes/winamp live.


From brian at freeswitch.org  Tue Dec  2 06:45:47 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 08:45:47 -0600
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20791453.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com>
Message-ID: <B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>

Can you show me the full XML for this extension including the regular  
expression?

/b

On Dec 2, 2008, at 7:25 AM, ccav wrote:

> transfer to voicemail is as follows
> <action application="answer"/>
> <action application="voicemail" data="default $${domain} $2"/>



From brian at freeswitch.org  Tue Dec  2 06:47:18 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 08:47:18 -0600
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
Message-ID: <375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>

Are you on SVN trunk or what rev are you trying to use?

/b

On Dec 2, 2008, at 7:48 AM, Dennis wrote:

> it seems, that fs has to stream to recording file to a streaming
> server (like icecast), right? but if we do "api uuid_record uuid start
> shout://user:passwd at ip-adress:12345/" (and other combinations), we get
> an error:
> 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid  
> URL: xxxxx
> 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851
> switch_ivr_record_session() Error opening shout://xxxx



From odermann at googlemail.com  Tue Dec  2 07:03:24 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 16:03:24 +0100
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
Message-ID: <5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>

i am using the latest svn trunk from today.


2008/12/2 Brian West <brian at freeswitch.org>:
> Are you on SVN trunk or what rev are you trying to use?
>
> /b
>
> On Dec 2, 2008, at 7:48 AM, Dennis wrote:
>
>> it seems, that fs has to stream to recording file to a streaming
>> server (like icecast), right? but if we do "api uuid_record uuid start
>> shout://user:passwd at ip-adress:12345/" (and other combinations), we get
>> an error:
>> 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid
>> URL: xxxxx
>> 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851
>> switch_ivr_record_session() Error opening shout://xxxx
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Dec  2 07:08:30 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 09:08:30 -0600
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
	<5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
Message-ID: <5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>

And you have your shoutcast/icecast server set up and functional?

/b

On Dec 2, 2008, at 9:03 AM, Dennis wrote:

> i am using the latest svn trunk from today.



From odermann at googlemail.com  Tue Dec  2 07:25:20 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 16:25:20 +0100
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
	<5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
	<5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
Message-ID: <5e414ed0812020725y138737beu54cc5d0808161093@mail.gmail.com>

no, not yet. i am still fiddling arround with icecast2.

we tried it with someone, who offers radiostreams. perhaps this just
works with icecast(2) and shoutcast?



2008/12/2 Brian West <brian at freeswitch.org>:
> And you have your shoutcast/icecast server set up and functional?
>
> /b
>
> On Dec 2, 2008, at 9:03 AM, Dennis wrote:
>
>> i am using the latest svn trunk from today.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Dec  2 07:34:39 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 09:34:39 -0600
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5e414ed0812020725y138737beu54cc5d0808161093@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
	<5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
	<5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
	<5e414ed0812020725y138737beu54cc5d0808161093@mail.gmail.com>
Message-ID: <A6D64AAE-DB9B-4206-B76B-81FE07F490AD@freeswitch.org>

icecast2 is a known working server we have talked to before.

/b

On Dec 2, 2008, at 9:25 AM, Dennis wrote:

> no, not yet. i am still fiddling arround with icecast2.
>
> we tried it with someone, who offers radiostreams. perhaps this just
> works with icecast(2) and shoutcast?



From anthony.minessale at gmail.com  Tue Dec  2 07:51:39 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 09:51:39 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <4934D197.8080007@3c.co.uk>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
Message-ID: <191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>

If you can get it to break on linux I will ssh in and fix it for you.
If you cannot, i can try to fix it for you over rdp but that won't be very
fun.

We can think about reinstating mod_lumenvox as well as another windows based
asr
alternative.  I deleted it for the same reason we will probably delete
mod_openmrcp because
nobody was using it and there was no way to support it because our dev
licenses had expired.

Lumenvox has offered us some new dev licenses to bring it back but I would
need someone to actually want it to work to put in charge of it.

We will be clear about what is supported and what is not in the 1.0.2
release scheduled
to be released in the near future.




On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:

>  Hi Anthony,
>
> mod_openmrcp was a contribution to the community by a 3rd party individual.
>
> As i have clearly stated in 2 previous emails, the man has decided to
> discontinue the openmrcp project.
> So now we are left with the remains of the module and discontinued code.
> This was not our decision it was his.
>
> I absolutely understand this but it's important, from a user point of view,
> to be able to know which bits of FS are current/supported and which aren't.
>
> Some people use it without issue which may mean that the crash you reported
> is windows specific and I do not have a working lab of any mrcp capbable
> system to try it against in unix for that matter.  I have a list of work to
> do from here to the moon and back so on an issue like this, unless someone
> can hand me login credentials to some box and give me a phone number to dial
> to reporduce the issue, it will be a long time until we can deal with it.
>
> It's useful to know that there are people using mod_openmrcp without issue:
> I did ask here if anyone was a while back, and no-one fessed up.  I'll give
> it a go on a Linux box and report back.  And if you'd like a dev/test
> environment set up, then just tell me which one.
>
> And the question arises, should we bother working on it anymore if the lib
> has been abandoned and we cannot even get any support from it's author which
> is where the problem most likely lies.
>
> I try not to get too annoyed by these remarks about what we *ought to do*
> because I know people lose sight of how much of the work to support the
> project is done by a small group of 3 people and not the 2000 people it
> appears to be from the outside looking in. (I've been answering email for 4
> hours now)
>
> Those guys who claim to have all that money in an offshore bank account are
> lying - you don't have to reply to them in future ;-)  Seriously, though, I
> don't think it's too outrageous an idea to document what's supported and
> were you (for example) to have suggested that I get in touch with the
> contributors to the various modules, ask them what their view of its status
> is, condense the answers in to a list and report back, it's something I'd
> quite happily do.
>
> My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
> FS that we can maintain in tree knowing the development can be supported by
> the original author.
>
> Quite happy to participate in that, too.. the problem is that I've a demo
> to do like yesterday and the timescale for mod_unimrcp is a bit on the long
> side for that.  I'd rather not have to do it with Asterisk and Lumenvox..!
>
> Cheers --
>
> Dave
>
>
>
> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>
>>  Hi Mike,
>>
>> My experience is that it's somewhat broken - it took two trivial tweaks to
>> get it to work with IBM's ASR and TTS, but there's a more intractable
>> problem to do with memory getting overwritten (I assume that this is
>> something to do with something being freed when it shouldn't be) which
>> causes a segfault on the second or third session after the module being
>> loaded.
>>
>> Without wishing to sound like a stuck record, one thing that you guys
>> really ought to do is to decide what's supported and what isn't, and make
>> this obvious - for example, move unsupported modules to a different place in
>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>> round in circles a bit trying to make it work, and then find out (a) that it
>> doesn't and (b) it's not going to be fixed because it's not supported.
>>
>> Cheers --
>>
>> Dave
>>
>>  I would not say it is totally broken, it is known to work in quite a few
>> places, but we are unlikely to be doing any new fixes in it.
>>  Mike
>>
>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>
>>  Hi Anthony,
>>
>> Oh! OK.
>>
>> So is this module "totally broken".
>>
>> I say this because I can't seem to get it to work at all with the example
>> in that Mod_openmrcp wiki page but I thought it might because I'm not be
>> using the right Cepstral software (freetrial download versus the paided for
>> SDK) or that I'm not using the right port numbers or something else I didn't
>> do. I used TcpView to look at local port associated with my Cepstral
>> software and changed a few things but still nothing. I changed the loglevel
>> setting to 7 in the wiki's example but I don't see the kind of output on the
>> console that I would expect for debug mode.
>>
>> Thanks. Mark.
>>
>>
>>    ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> David Knell, Director, 3C Limited
> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Tue Dec  2 07:54:57 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 09:54:57 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <8CB2262CD41B5B4-430-201D@WEBMAIL-MA12.sysops.aol.com>
References: <8CB2262CD41B5B4-430-201D@WEBMAIL-MA12.sysops.aol.com>
Message-ID: <191c3a030812020754t26951be6i904fcf19f38a02d9@mail.gmail.com>

FreeSWITCH has an enterprise scale SIP UA.  Not only can it listen on other
ports it can
listen and work on as many ip:port combos as you want simultaneously each
with it's own specific config.

If you have an affinity for port 5060 you can always bring up 2 IP on the
same box and give one to each application.  You can essentially do whatever
you want. It's your box and everything involved is configurable.




On Tue, Dec 2, 2008 at 1:00 AM, <mszlazak at aol.com> wrote:

>  I need to barge in again and add to my last post with this email from
> Voxeo support. Here is their response to the port binding conflict and it
> brings up a possible problem if FreeSwitch will be looking for Prophecy at
> that port? I assumed it would if I set up the extension right but now I
> don't know and need your assistance with this issue ... as well.
>
> Thank you.
>
> MESSAGE:
>
> Hi Mark,
>
>
> You are correct in that having multiple applications binding to the same port
>
> can cause a bundle of problems. You can configure Prophecy to stay away from
>
> port 5060, but then the question is whether FreeSwitch will be looking for
>
> Prophecy at that port (if its assuming that it's residing on a different box).
>
> Port 5060 is the standard for SIP traffic.
>
>
> To get Prophecy off 5060 you will need to edit the config.xml and
>
> callrouting.xml files. You will need to search out all instances of "5060" and
>
> replace with, perhaps, port 5068. For instance:
>
>
> <item name="ListenOnIP1">0.0.0.0:5068</item>
>
> <item name="ListenOnIP2">0.0.0.0:5061</item>
>
> <item name="ListenOnIP3">0.0.0.0:5067</item>
>
> <item name="ListenOnIP4">0.0.0.0:5063</item>
>
> <item name="ListenOnIP5">0.0.0.0:5064</item>
>
> <item name="ListenOnIP6">0.0.0.0:5065</item>
>
>
> instead of this...
>
>
> <item name="ListenOnIP1">0.0.0.0:5060</item>
>
> <item name="ListenOnIP2">0.0.0.0:5061</item>
>
> <item name="ListenOnIP3">0.0.0.0:5062</item>
>
> <item name="ListenOnIP4">0.0.0.0:5063</item>
>
> <item name="ListenOnIP5">0.0.0.0:5064</item>
>
> <item name="ListenOnIP6">0.0.0.0:5065</item>
>
>
> Regards,
>
> Jeff Kustermann
>
> Voxeo Support
>
>
>
>
>
>  -----Original Message-----
> From: mszlazak at aol.com
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 1 Dec 2008 10:40 pm
> Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP
>
>
>  Just to follow up.
>
> Moshe Yudkowsky has an article on "Routing calls from FreeSwitch to
> Prophecy":  http://www.prophecy2006.com/node/145
>
> My problem is that Freeswitch and Prophecy need to be on the same machine
> BUT both need to bind to port 5060 so I'm getting errors from one or the
> other depending who's running first.
>
> So can I change what port(s) FS uses and that way avoid this conflict?
> Maybe, this might let me bridge the call via FreeSwitch to Prophecy similar
> to what Moshe's article discusses???
>
>  -----Original Message-----
> From: mszlazak at aol.com
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 1 Dec 2008 4:44 pm
> Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP
>
>
>  Does "bridging" a call from FS to Voxeo's Prophecy server require
> openMRCP? If not then the other issue I might have is a database look up
> that is part of the dialogue that maybe need as the person response to
> prompts from the asr. It's possible to run a php script for the database
> stuff that Prophecy might need or could that happen via Javascript in FS?
> Then after the dialogue has completed I go from Prophecy back to FS.
>
>  -----Original Message-----
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 1 Dec 2008 11:17 am
> Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP
>
>  mod_openmrcp was a contribution to the community by a 3rd party
> individual.
>
> As i have clearly stated in 2 previous emails, the man has decided to
> discontinue the openmrcp project.
> So now we are left with the remains of the module and discontinued code.
> This was not our decision it was his.
>
> Since the author of openmrcp has stated that he has a new unimrcp we are
> certainly going to
> work towards getting mod_unimrcp to replace mod_openmrcp.  He had already
> commented on that previous thread to state he is willing to consider making
> a new module.
>
> Some people use it without issue which may mean that the crash you reported
> is windows specific and I do not have a working lab of any mrcp capbable
> system to try it against in unix for that matter.  I have a list of work to
> do from here to the moon and back so on an issue like this, unless someone
> can hand me login credentials to some box and give me a phone number to dial
> to reporduce the issue, it will be a long time until we can deal with it.
> And the question arises, should we bother working on it anymore if the lib
> has been abandoned and we cannot even get any support from it's author which
> is where the problem most likely lies.
>
> I try not to get too annoyed by these remarks about what we *ought to do*
> because I know people lose sight of how much of the work to support the
> project is done by a small group of 3 people and not the 2000 people it
> appears to be from the outside looking in. (I've been answering email for 4
> hours now)
>
> My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
> FS that we can maintain in tree knowing the development can be supported by
> the original author.
>
>
> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>
>>  Hi Mike,
>>
>> My experience is that it's somewhat broken - it took two trivial tweaks to
>> get it to work with IBM's ASR and TTS, but there's a more intractable
>> problem to do with memory getting overwritten (I assume that this is
>> something to do with something being freed when it shouldn't be) which
>> causes a segfault on the second or third session after the module being
>> loaded.
>>
>> Without wishing to sound like a stuck record, one thing that you guys
>> really ought to do is to decide what's supported and what isn't, and make
>> this obvious - for example, move unsupported modules to a different place in
>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>> round in circles a bit trying to make it work, and then find out (a) that it
>> doesn't and (b) it's not going to be fixed because it's not supported.
>>
>> Cheers --
>>
>> Dave
>>
>>  I would not say it is totally broken, it is known to work in quite a few
>> places, but we are unlikely to be doing any new fixes in it.
>>  Mike
>>
>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>
>>  Hi Anthony,
>>
>> Oh! OK.
>>
>> So is this module "totally broken".
>>
>> I say this because I can't seem to get it to work at all with the example
>> in that Mod_openmrcp wiki page but I thought it might because I'm not be
>> using the right Cepstral software (freetrial download versus the paided for
>> SDK) or that I'm not using the right port numbers or something else I didn't
>> do. I used TcpView to look at local port associated with my Cepstral
>> software and changed a few things but still nothing. I changed the loglevel
>> setting to 7 in the wiki's example but I don't see the kind of output on the
>> console that I would expect for debug mode.
>>
>> Thanks. Mark.
>>
>>
>>    ------------------------------
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> --
>>
>>
>>
>>
>>
>>
>>
>>
>> David Knell, Director, 3C Limited
>>
>>
>>
>>
>>
>>
>>
>>
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>>
>>
>>
>>
>>
>>
>>
>> http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
>
>
>
>
>
>
>
>
> Freeswitch-users mailing list
>
>
>
>
>
>
>
> Freeswitch-users at lists.freeswitch.org
>
>
>
>
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
>
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
>
>
>
>
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
> ------------------------------
> Tis the season to save your money! Get the new AOL Holiday Toolbar<http://toolbar.aol.com/holiday/download.html?ncid=emlweusdown00000008>for money saving offers and gift ideas.
>
> _______________________________________________
>
>
>
>
> Freeswitch-users mailing list
>
>
>
> Freeswitch-users at lists.freeswitch.org
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
> http://www.freeswitch.org
>
>
>
>
> ------------------------------
> Tis the season to save your money! Get the new AOL Holiday Toolbar<http://toolbar.aol.com/holiday/download.html?ncid=emlweusdown00000008>for money saving offers and gift ideas.
>
> _______________________________________________
>
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> ------------------------------
> Tis the season to save your money! Get the new AOL Holiday Toolbar<http://toolbar.aol.com/holiday/download.html?ncid=emlweusdown00000008>for money saving offers and gift ideas.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Tue Dec  2 08:07:21 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 17:07:21 +0100
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
	<5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
	<5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
Message-ID: <5e414ed0812020807n77f4b3d3pd9cd18d72e416029@mail.gmail.com>

sorry, problem solved :-)

it works very good with icecast2.



2008/12/2 Brian West <brian at freeswitch.org>:
> And you have your shoutcast/icecast server set up and functional?
>
> /b
>
> On Dec 2, 2008, at 9:03 AM, Dennis wrote:
>
>> i am using the latest svn trunk from today.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Tue Dec  2 08:09:09 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 10:09:09 -0600
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
Message-ID: <191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>

from the source tree of FS please type

"make current"

when it completes, retest the call.



On Tue, Dec 2, 2008 at 5:07 AM, Baskar <yudha2008 at gmail.com> wrote:

> *Hi,
>
> This is the svn version i have installed before a month
>
> FreeSWITCH Version 1.0.trunk (10130M)
> *
> --
> *Warm Regards,
> N.Baskar*
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From carlos.talbot at gmail.com  Tue Dec  2 08:13:28 2008
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 2 Dec 2008 10:13:28 -0600
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <000001c9530d$912d86d0$b3889470$@com>
References: <000001c9530d$912d86d0$b3889470$@com>
Message-ID: <5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>

Have you tried the latest msi build? It's based off svn 10564.

Carlos

On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:

> I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
> on a Windows Vista machine using the precompiled .msi - actually the same
> machine).
>
> Using the default configuration files, and using 2 Snom 360 phones I dialed
> from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
> instantly, but under Windows it takes 1-2 seconds before it starts ringing.
>
> It seems to be in the dialplan the time is spent. From the time I see this
> line on the console:
>
> [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
> context default
>
> Until the next thing happens it always takes at least 1 full second, but on
> the Mac it happens instantly.
>
> Why is the Windows build this much slower? Is it a known problem?
>
> I get the feeling that the majority of the FS community is Unix based,
> which
> is fine by me, but I would really like to know just how well supported and
> stable the Win32 build is and if this is currently a viable way to go, or
> if
> I should stick to Linux/BSD/Mac for production use?
>
>
> // Per
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stkn at freeswitch.org  Tue Dec  2 08:20:16 2008
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Tue, 2 Dec 2008 17:20:16 +0100
Subject: [Freeswitch-users] Support for Junghanns duoBRI
In-Reply-To: <B22D2E3D-9C6F-43C6-A634-DC00B91BDBBB@jerris.com>
References: <49340242.3040403@gmail.com>
	<B22D2E3D-9C6F-43C6-A634-DC00B91BDBBB@jerris.com>
Message-ID: <200812021720.16188.stkn@freeswitch.org>

All HFC-based cards supported by bristuffed Zaptel should work.

Stefan

Am Monday 01 December 2008 schrieb Michael Jerris:
> The bri support is still in development, basic calls on ptmp bri do  
> appear to work, although I am not sure with what hardware.
> 
> Mike
> 
> 
> On Dec 1, 2008, at 10:26 AM, Sergey Kirillov wrote:
> 
> > Greetings,
> >
> > Can somebody tell me, if it is possible to use duoBRI card
> > (http://www.junghanns.net/en/duobri_express_produkt.html) from
> > Junghanns.net together with Freeswitch?
> >
> > I've found that this card has Zaptel drivers, and Freeswitch has
> > mod_openzap. On the other side, I saw somewhere in wiki that  
> > Freeswitch
> > does not support BRI at all at the moment.
> >
> >
> > Please confirm or allay my apprehensions.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


-- 
Stefan Knoblich
Systemadministrator

axsentis GmbH
Eupener Strasse 74
50933 K?ln

Tel: 0180 - 506 705 521*
Fax: 0180 - 506 705 529*

E-Mail: s.knoblich at axsentis.de
Web: www.axsentis.de

Eingetragen beim AG K?ln: HR B 56238
UST-ID: DE244977565
Gesellschafter-Gesch?ftsf?hrer: 
Yan Lecomte, Eduard Schlein, Apostolos Varsamis

*14ct/min aus dem Festnetz der T-Com | dtms


From gilbertandrew at me.com  Tue Dec  2 08:24:48 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 02 Dec 2008 11:24:48 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
	<191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
Message-ID: <39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>

Mark and David,

I am willing to help some with testing here as well, if you need it.  
Ping me directly or we can get on the IRC. I am on Mac OS, but have  
readily available vm's with Debian, etc. I also have Prophecy.

I have a general interest in an ASR solution as well. Voxeo is great,  
but using it as an MRCP proxy seems odd. As a full fledged VXML  
solution it is great, if you can afford it. But having a good ASR  
solution is good first step to trying to get something like OpenVXI  
working as well.

That said, seems like a bounty or money to help FS is a better spend  
anyway. It is a one time cost, not a variable cost. And it goes  
straight to the guys doing the real work.

I built unimrcp last night, it was quite straight forward. In theory,  
if I weren't old and my C/autoconf skills rather atrophied, it  
wouldn't seem like it would be that huge a deal to port/fix openmrcp  
to unimrcp.

Finally, Anthony I was looking at the Lumenvox path as well, but got  
deterred by the licensing hassle. This seems to be a universal ASR  
issue. I would reason I can find the old module in SVN? Were they  
going to grant "community dev" licenses? Again - I am willing to  
volunteer to do some testing/doc at least.

Andy



On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote:

> If you can get it to break on linux I will ssh in and fix it for you.
> If you cannot, i can try to fix it for you over rdp but that won't  
> be very fun.
>
> We can think about reinstating mod_lumenvox as well as another  
> windows based asr
> alternative.  I deleted it for the same reason we will probably  
> delete mod_openmrcp because
> nobody was using it and there was no way to support it because our  
> dev licenses had expired.
>
> Lumenvox has offered us some new dev licenses to bring it back but I  
> would need someone to actually want it to work to put in charge of it.
>
> We will be clear about what is supported and what is not in the  
> 1.0.2 release scheduled
> to be released in the near future.
>
>
>
>
> On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:
> Hi Anthony,
>
>> mod_openmrcp was a contribution to the community by a 3rd party  
>> individual.
>>
>> As i have clearly stated in 2 previous emails, the man has decided  
>> to discontinue the openmrcp project.
>> So now we are left with the remains of the module and discontinued  
>> code.  This was not our decision it was his.
> I absolutely understand this but it's important, from a user point  
> of view, to be able to know which bits of FS are current/supported  
> and which aren't.
>
>> Some people use it without issue which may mean that the crash you  
>> reported is windows specific and I do not have a working lab of any  
>> mrcp capbable system to try it against in unix for that matter.  I  
>> have a list of work to do from here to the moon and back so on an  
>> issue like this, unless someone can hand me login credentials to  
>> some box and give me a phone number to dial to reporduce the issue,  
>> it will be a long time until we can deal with it.
> It's useful to know that there are people using mod_openmrcp without  
> issue: I did ask here if anyone was a while back, and no-one fessed  
> up.  I'll give it a go on a Linux box and report back.  And if you'd  
> like a dev/test environment set up, then just tell me which one.
>
>> And the question arises, should we bother working on it anymore if  
>> the lib has been abandoned and we cannot even get any support from  
>> it's author which is where the problem most likely lies.
>>
>> I try not to get too annoyed by these remarks about what we *ought  
>> to do* because I know people lose sight of how much of the work to  
>> support the project is done by a small group of 3 people and not  
>> the 2000 people it appears to be from the outside looking in. (I've  
>> been answering email for 4 hours now)
> Those guys who claim to have all that money in an offshore bank  
> account are lying - you don't have to reply to them in future ;-)   
> Seriously, though, I don't think it's too outrageous an idea to  
> document what's supported and were you (for example) to have  
> suggested that I get in touch with the contributors to the various  
> modules, ask them what their view of its status is, condense the  
> answers in to a list and report back, it's something I'd quite  
> happily do.
>
>> My suggestion is to pool some cash and pay the guy to make  
>> mod_unimrcp for FS that we can maintain in tree knowing the  
>> development can be supported by the original author.
> Quite happy to participate in that, too.. the problem is that I've a  
> demo to do like yesterday and the timescale for mod_unimrcp is a bit  
> on the long side for that.  I'd rather not have to do it with  
> Asterisk and Lumenvox..!
>
> Cheers --
>
> Dave
>
>>
>>
>> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>> Hi Mike,
>>
>> My experience is that it's somewhat broken - it took two trivial  
>> tweaks to get it to work with IBM's ASR and TTS, but there's a more  
>> intractable problem to do with memory getting overwritten (I assume  
>> that this is something to do with something being freed when it  
>> shouldn't be) which causes a segfault on the second or third  
>> session after the module being loaded.
>>
>> Without wishing to sound like a stuck record, one thing that you  
>> guys really ought to do is to decide what's supported and what  
>> isn't, and make this obvious - for example, move unsupported  
>> modules to a different place in the tree, don't have them built by  
>> default, etc.  MRCP is in the specsheet on the Wiki.  Otherwise  
>> folk like Mark and I spend time installing stuff, go round in  
>> circles a bit trying to make it work, and then find out (a) that it  
>> doesn't and (b) it's not going to be fixed because it's not  
>> supported.
>>
>> Cheers --
>>
>> Dave
>>> I would not say it is totally broken, it is known to work in quite  
>>> a few places, but we are unlikely to be doing any new fixes in it.
>>>
>>> Mike
>>>
>>> On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>>
>>>> Hi Anthony,
>>>>
>>>> Oh! OK.
>>>>
>>>> So is this module "totally broken".
>>>>
>>>> I say this because I can't seem to get it to work at all with the  
>>>> example in that Mod_openmrcp wiki page but I thought it might  
>>>> because I'm not be using the right Cepstral software (freetrial  
>>>> download versus the paided for SDK) or that I'm not using the  
>>>> right port numbers or something else I didn't do. I used TcpView  
>>>> to look at local port associated with my Cepstral software and  
>>>> changed a few things but still nothing. I changed the loglevel  
>>>> setting to 7 in the wiki's example but I don't see the kind of  
>>>> output on the console that I would expect for debug mode.
>>>>
>>>> Thanks. Mark.
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> -- 
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>> http://www.3c.co.uk
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> -- 
> David Knell, Director, 3C Limited
> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
> http://www.3c.co.uk
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Dec  2 08:32:31 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 10:32:31 -0600
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
Message-ID: <191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>

Naturally, either way is stupid.

The whole idea of putting the transport in a uri param is equally stupid to
using 2 different protocol names but since SIP is the descendant of http it
they decided to stick with the stupidity of http/https and have sip/sips
which is almost as if it was designed to break all software trying to keep
up with url syntax.

If they are going to insist on using text params you'd think something like
transport=foo;security=tls would be even *more* flexable in case alternate
methods to encrypt crop up.

This is, of course, the first step into a lengthy 12 hour discussion on how
stupid SIP and url/text based
protocols are.

I dare someone to crank up the pcap on a box doing SIP presence for 20
phones and "read"
the 1200 byte messages with all kinds of hyeroglyphic url syntax and
embedded xml payloads and write
up a paper on how much "sense" it makes to have it be "readable".

PS

supposedly sofia can support sctp,
someone should try it.



On Mon, Dec 1, 2008 at 9:43 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On 12/1/08, Thomas Troy <ttroy50 at gmail.com> wrote:
> ..snip..
> >
> > Out of interest do you have any links to anywhere this is discussed in
> terms
> > of general sip implementations?
> >
>
> Uh oh, here we go again...
>
> http://www.iana.org/assignments/sip-parameters
> http://tools.ietf.org/html/rfc3969
>
>
> https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-August/010047.html
>
> Implementation wise, most devices tend to use transport=tls:
>
> SIPFoundry - From what I've seen
> Snom
> SERs
> Asterisk (If you are using TLS)
> Cisco - I *believe* you can use either a SIPS URI or the transport=tls
> parameter for various SIP targets
>
>  As the RFC (basically) states (RFC3261, section 12.1.x),
> transport=tls was deprecated in RFC 3261 because you should also be
> able to do TLS over SCTP (RFC3436), which makes transport=tls a bit
> ambiguous. sips:user at domain;transport=tcp or
> sips:user at domain;transport=sctp is a bit more flexible.
>
>  I don't know if I've ever seen anything default to SIPS URIs.  I
> also don't think I've ever specifically tried using them.  However, my
> experience with TLS is admittedly somewhat limited so this shouldn't
> be taken as gospel.  As you can see from the discussions on
> sip-implementors, this gets interesting when different devices are
> traversing a proxy using different URI schemes...
>
>  However, I suspect this won't become an issue until most SIP
> implementations support SCTP.  That should be exciting! ;)
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Tue Dec  2 08:43:44 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 10:43:44 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
	<191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
	<39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>
Message-ID: <191c3a030812020843m1bed6ab2mf77f1132ec1f26fa@mail.gmail.com>

from build root:

svn co -r8809
http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvoxsrc/mod/asr_tts/mod_lumenvox


They did seem to express an interest in granting some dev licenses when they
realized we took the code out of tree but I have not actually dealt with the
issue yet because I have been overwhelmed.

I don't know if this code works anymore with the latest revision of the api
but there it is.





On Tue, Dec 2, 2008 at 10:24 AM, Andrew Gilbert <gilbertandrew at me.com>wrote:

> Mark and David,
>
> I am willing to help some with testing here as well, if you need it. Ping
> me directly or we can get on the IRC. I am on Mac OS, but have readily
> available vm's with Debian, etc. I also have Prophecy.
>
> I have a general interest in an ASR solution as well. Voxeo is great, but
> using it as an MRCP proxy seems odd. As a full fledged VXML solution it is
> great, if you can afford it. But having a good ASR solution is good first
> step to trying to get something like OpenVXI working as well.
>
> That said, seems like a bounty or money to help FS is a better spend
> anyway. It is a one time cost, not a variable cost. And it goes straight to
> the guys doing the real work.
>
> I built unimrcp last night, it was quite straight forward. In theory, if I
> weren't old and my C/autoconf skills rather atrophied, it wouldn't seem like
> it would be that huge a deal to port/fix openmrcp to unimrcp.
>
> Finally, Anthony I was looking at the Lumenvox path as well, but got
> deterred by the licensing hassle. This seems to be a universal ASR issue. I
> would reason I can find the old module in SVN? Were they going to grant
> "community dev" licenses? Again - I am willing to volunteer to do some
> testing/doc at least.
>
> Andy
>
>
>
> On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote:
>
> If you can get it to break on linux I will ssh in and fix it for you.
> If you cannot, i can try to fix it for you over rdp but that won't be very
> fun.
>
> We can think about reinstating mod_lumenvox as well as another windows
> based asr
> alternative.  I deleted it for the same reason we will probably delete
> mod_openmrcp because
> nobody was using it and there was no way to support it because our dev
> licenses had expired.
>
> Lumenvox has offered us some new dev licenses to bring it back but I would
> need someone to actually want it to work to put in charge of it.
>
> We will be clear about what is supported and what is not in the 1.0.2
> release scheduled
> to be released in the near future.
>
>
>
>
> On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:
>
>>  Hi Anthony,
>>
>> mod_openmrcp was a contribution to the community by a 3rd party
>> individual.
>>
>> As i have clearly stated in 2 previous emails, the man has decided to
>> discontinue the openmrcp project.
>> So now we are left with the remains of the module and discontinued code.
>> This was not our decision it was his.
>>
>> I absolutely understand this but it's important, from a user point of
>> view, to be able to know which bits of FS are current/supported and which
>> aren't.
>>
>> Some people use it without issue which may mean that the crash you
>> reported is windows specific and I do not have a working lab of any mrcp
>> capbable system to try it against in unix for that matter.  I have a list of
>> work to do from here to the moon and back so on an issue like this, unless
>> someone can hand me login credentials to some box and give me a phone number
>> to dial to reporduce the issue, it will be a long time until we can deal
>> with it.
>>
>> It's useful to know that there are people using mod_openmrcp without
>> issue: I did ask here if anyone was a while back, and no-one fessed up.
>> I'll give it a go on a Linux box and report back.  And if you'd like a
>> dev/test environment set up, then just tell me which one.
>>
>> And the question arises, should we bother working on it anymore if the lib
>> has been abandoned and we cannot even get any support from it's author which
>> is where the problem most likely lies.
>>
>> I try not to get too annoyed by these remarks about what we *ought to do*
>> because I know people lose sight of how much of the work to support the
>> project is done by a small group of 3 people and not the 2000 people it
>> appears to be from the outside looking in. (I've been answering email for 4
>> hours now)
>>
>> Those guys who claim to have all that money in an offshore bank account
>> are lying - you don't have to reply to them in future ;-)  Seriously,
>> though, I don't think it's too outrageous an idea to document what's
>> supported and were you (for example) to have suggested that I get in touch
>> with the contributors to the various modules, ask them what their view of
>> its status is, condense the answers in to a list and report back, it's
>> something I'd quite happily do.
>>
>> My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
>> FS that we can maintain in tree knowing the development can be supported by
>> the original author.
>>
>> Quite happy to participate in that, too.. the problem is that I've a demo
>> to do like yesterday and the timescale for mod_unimrcp is a bit on the long
>> side for that.  I'd rather not have to do it with Asterisk and Lumenvox..!
>>
>> Cheers --
>>
>> Dave
>>
>>
>>
>> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>>
>>>  Hi Mike,
>>>
>>> My experience is that it's somewhat broken - it took two trivial tweaks
>>> to get it to work with IBM's ASR and TTS, but there's a more intractable
>>> problem to do with memory getting overwritten (I assume that this is
>>> something to do with something being freed when it shouldn't be) which
>>> causes a segfault on the second or third session after the module being
>>> loaded.
>>>
>>> Without wishing to sound like a stuck record, one thing that you guys
>>> really ought to do is to decide what's supported and what isn't, and make
>>> this obvious - for example, move unsupported modules to a different place in
>>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>>> round in circles a bit trying to make it work, and then find out (a) that it
>>> doesn't and (b) it's not going to be fixed because it's not supported.
>>>
>>> Cheers --
>>>
>>> Dave
>>>
>>>  I would not say it is totally broken, it is known to work in quite a
>>> few places, but we are unlikely to be doing any new fixes in it.
>>>  Mike
>>>
>>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>>
>>>  Hi Anthony,
>>>
>>> Oh! OK.
>>>
>>> So is this module "totally broken".
>>>
>>> I say this because I can't seem to get it to work at all with the example
>>> in that Mod_openmrcp wiki page but I thought it might because I'm not be
>>> using the right Cepstral software (freetrial download versus the paided for
>>> SDK) or that I'm not using the right port numbers or something else I didn't
>>> do. I used TcpView to look at local port associated with my Cepstral
>>> software and changed a few things but still nothing. I changed the loglevel
>>> setting to 7 in the wiki's example but I don't see the kind of output on the
>>> console that I would expect for debug mode.
>>>
>>> Thanks. Mark.
>>>
>>>
>>>    ------------------------------
>>> _______________________________________________
>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> David Knell, Director, 3C Limited
>>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> ------------------------------
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gilbertandrew at me.com  Tue Dec  2 09:27:25 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 02 Dec 2008 12:27:25 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812020843m1bed6ab2mf77f1132ec1f26fa@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
	<191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
	<39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>
	<191c3a030812020843m1bed6ab2mf77f1132ec1f26fa@mail.gmail.com>
Message-ID: <7ED7E135-711F-4E7F-BB3B-6B6014211B90@me.com>

Ok

I have a ping in with Lumenvox about dev licensing, and pulled the  
mod. Not sure where this will go, but will take a peek at things.  
Balancing the effort against something like getting unimcrp going and/ 
or openmrcp tested and stable.

Thanks.

Andy


On Dec 2, 2008, at 11:43 AM, Anthony Minessale wrote:

> from build root:
>
> svn co -r8809 http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvox 
>  src/mod/asr_tts/mod_lumenvox
>
>
> They did seem to express an interest in granting some dev licenses  
> when they realized we took the code out of tree but I have not  
> actually dealt with the issue yet because I have been overwhelmed.
>
> I don't know if this code works anymore with the latest revision of  
> the api but there it is.
>
>
>
>
>
> On Tue, Dec 2, 2008 at 10:24 AM, Andrew Gilbert  
> <gilbertandrew at me.com> wrote:
> Mark and David,
>
> I am willing to help some with testing here as well, if you need it.  
> Ping me directly or we can get on the IRC. I am on Mac OS, but have  
> readily available vm's with Debian, etc. I also have Prophecy.
>
> I have a general interest in an ASR solution as well. Voxeo is  
> great, but using it as an MRCP proxy seems odd. As a full fledged  
> VXML solution it is great, if you can afford it. But having a good  
> ASR solution is good first step to trying to get something like  
> OpenVXI working as well.
>
> That said, seems like a bounty or money to help FS is a better spend  
> anyway. It is a one time cost, not a variable cost. And it goes  
> straight to the guys doing the real work.
>
> I built unimrcp last night, it was quite straight forward. In  
> theory, if I weren't old and my C/autoconf skills rather atrophied,  
> it wouldn't seem like it would be that huge a deal to port/fix  
> openmrcp to unimrcp.
>
> Finally, Anthony I was looking at the Lumenvox path as well, but got  
> deterred by the licensing hassle. This seems to be a universal ASR  
> issue. I would reason I can find the old module in SVN? Were they  
> going to grant "community dev" licenses? Again - I am willing to  
> volunteer to do some testing/doc at least.
>
> Andy
>
>
>
> On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote:
>
>> If you can get it to break on linux I will ssh in and fix it for you.
>> If you cannot, i can try to fix it for you over rdp but that won't  
>> be very fun.
>>
>> We can think about reinstating mod_lumenvox as well as another  
>> windows based asr
>> alternative.  I deleted it for the same reason we will probably  
>> delete mod_openmrcp because
>> nobody was using it and there was no way to support it because our  
>> dev licenses had expired.
>>
>> Lumenvox has offered us some new dev licenses to bring it back but  
>> I would need someone to actually want it to work to put in charge  
>> of it.
>>
>> We will be clear about what is supported and what is not in the  
>> 1.0.2 release scheduled
>> to be released in the near future.
>>
>>
>>
>>
>> On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:
>> Hi Anthony,
>>
>>> mod_openmrcp was a contribution to the community by a 3rd party  
>>> individual.
>>>
>>> As i have clearly stated in 2 previous emails, the man has decided  
>>> to discontinue the openmrcp project.
>>> So now we are left with the remains of the module and discontinued  
>>> code.  This was not our decision it was his.
>> I absolutely understand this but it's important, from a user point  
>> of view, to be able to know which bits of FS are current/supported  
>> and which aren't.
>>
>>> Some people use it without issue which may mean that the crash you  
>>> reported is windows specific and I do not have a working lab of  
>>> any mrcp capbable system to try it against in unix for that  
>>> matter.  I have a list of work to do from here to the moon and  
>>> back so on an issue like this, unless someone can hand me login  
>>> credentials to some box and give me a phone number to dial to  
>>> reporduce the issue, it will be a long time until we can deal with  
>>> it.
>> It's useful to know that there are people using mod_openmrcp  
>> without issue: I did ask here if anyone was a while back, and no- 
>> one fessed up.  I'll give it a go on a Linux box and report back.   
>> And if you'd like a dev/test environment set up, then just tell me  
>> which one.
>>
>>> And the question arises, should we bother working on it anymore if  
>>> the lib has been abandoned and we cannot even get any support from  
>>> it's author which is where the problem most likely lies.
>>>
>>> I try not to get too annoyed by these remarks about what we *ought  
>>> to do* because I know people lose sight of how much of the work to  
>>> support the project is done by a small group of 3 people and not  
>>> the 2000 people it appears to be from the outside looking in.  
>>> (I've been answering email for 4 hours now)
>> Those guys who claim to have all that money in an offshore bank  
>> account are lying - you don't have to reply to them in future ;-)   
>> Seriously, though, I don't think it's too outrageous an idea to  
>> document what's supported and were you (for example) to have  
>> suggested that I get in touch with the contributors to the various  
>> modules, ask them what their view of its status is, condense the  
>> answers in to a list and report back, it's something I'd quite  
>> happily do.
>>
>>> My suggestion is to pool some cash and pay the guy to make  
>>> mod_unimrcp for FS that we can maintain in tree knowing the  
>>> development can be supported by the original author.
>> Quite happy to participate in that, too.. the problem is that I've  
>> a demo to do like yesterday and the timescale for mod_unimrcp is a  
>> bit on the long side for that.  I'd rather not have to do it with  
>> Asterisk and Lumenvox..!
>>
>> Cheers --
>>
>> Dave
>>
>>>
>>>
>>> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>>> Hi Mike,
>>>
>>> My experience is that it's somewhat broken - it took two trivial  
>>> tweaks to get it to work with IBM's ASR and TTS, but there's a  
>>> more intractable problem to do with memory getting overwritten (I  
>>> assume that this is something to do with something being freed  
>>> when it shouldn't be) which causes a segfault on the second or  
>>> third session after the module being loaded.
>>>
>>> Without wishing to sound like a stuck record, one thing that you  
>>> guys really ought to do is to decide what's supported and what  
>>> isn't, and make this obvious - for example, move unsupported  
>>> modules to a different place in the tree, don't have them built by  
>>> default, etc.  MRCP is in the specsheet on the Wiki.  Otherwise  
>>> folk like Mark and I spend time installing stuff, go round in  
>>> circles a bit trying to make it work, and then find out (a) that  
>>> it doesn't and (b) it's not going to be fixed because it's not  
>>> supported.
>>>
>>> Cheers --
>>>
>>> Dave
>>>> I would not say it is totally broken, it is known to work in  
>>>> quite a few places, but we are unlikely to be doing any new fixes  
>>>> in it.
>>>>
>>>> Mike
>>>>
>>>> On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>>>
>>>>> Hi Anthony,
>>>>>
>>>>> Oh! OK.
>>>>>
>>>>> So is this module "totally broken".
>>>>>
>>>>> I say this because I can't seem to get it to work at all with  
>>>>> the example in that Mod_openmrcp wiki page but I thought it  
>>>>> might because I'm not be using the right Cepstral software  
>>>>> (freetrial download versus the paided for SDK) or that I'm not  
>>>>> using the right port numbers or something else I didn't do. I  
>>>>> used TcpView to look at local port associated with my Cepstral  
>>>>> software and changed a few things but still nothing. I changed  
>>>>> the loglevel setting to 7 in the wiki's example but I don't see  
>>>>> the kind of output on the console that I would expect for debug  
>>>>> mode.
>>>>>
>>>>> Thanks. Mark.
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> -- 
>>> David Knell, Director, 3C Limited
>>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>>> http://www.3c.co.uk
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> -- 
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:213-799-1400
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> -- 
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>> http://www.3c.co.uk
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From odermann at googlemail.com  Tue Dec  2 09:40:45 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 18:40:45 +0100
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
Message-ID: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>

hi,

because we do not get tired of testing and playing a lot with the
beloved fs, we now arrived at the fax feature :-)

i am not sure if the docs are up to date or if there was a lot of
development in the meantime. therefore i would like to ask, what is
possible and what will come in the near future.

we are using fs, socket outbound and php and would like to make
something like fax to mail as an additional service.

is t38 supported?
can i pass incoming faxes over the same socket as calls?
can i convert faxes into pdf?
is fax over sip reliable (as far as i have heard, under asterisk fax
is nothing one should use)?
and so on, and so on....

i would be very happy to hear some user experiences with fs and fax.
if it seems, that we can use fax with over socket outbound, we will do
hardcore testing ;-)

thanks,
dennis


From anthony.minessale at gmail.com  Tue Dec  2 09:58:59 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 11:58:59 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <7ED7E135-711F-4E7F-BB3B-6B6014211B90@me.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
	<191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
	<39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>
	<191c3a030812020843m1bed6ab2mf77f1132ec1f26fa@mail.gmail.com>
	<7ED7E135-711F-4E7F-BB3B-6B6014211B90@me.com>
Message-ID: <191c3a030812020958s30d0e6d2ub3452e7e63fa19d9@mail.gmail.com>

They contacted us shortly thereafter and asked if we want to have them sell
you the license for 50 bucks.
hmm, i wonder why i deleted the module.....

I will tell them that if they give you a developer license you will work on
getting it back into trunk.

On Tue, Dec 2, 2008 at 11:27 AM, Andrew Gilbert <gilbertandrew at me.com>wrote:

> Ok
>
> I have a ping in with Lumenvox about dev licensing, and pulled the mod. Not
> sure where this will go, but will take a peek at things. Balancing the
> effort against something like getting unimcrp going and/or openmrcp tested
> and stable.
>
> Thanks.
>
> Andy
>
>
> On Dec 2, 2008, at 11:43 AM, Anthony Minessale wrote:
>
> from build root:
>
> svn co -r8809
> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvoxsrc/mod/asr_tts/mod_lumenvox
>
>
> They did seem to express an interest in granting some dev licenses when
> they realized we took the code out of tree but I have not actually dealt
> with the issue yet because I have been overwhelmed.
>
> I don't know if this code works anymore with the latest revision of the api
> but there it is.
>
>
>
>
>
> On Tue, Dec 2, 2008 at 10:24 AM, Andrew Gilbert <gilbertandrew at me.com>wrote:
>
>> Mark and David,
>>
>> I am willing to help some with testing here as well, if you need it. Ping
>> me directly or we can get on the IRC. I am on Mac OS, but have readily
>> available vm's with Debian, etc. I also have Prophecy.
>>
>> I have a general interest in an ASR solution as well. Voxeo is great, but
>> using it as an MRCP proxy seems odd. As a full fledged VXML solution it is
>> great, if you can afford it. But having a good ASR solution is good first
>> step to trying to get something like OpenVXI working as well.
>>
>> That said, seems like a bounty or money to help FS is a better spend
>> anyway. It is a one time cost, not a variable cost. And it goes straight to
>> the guys doing the real work.
>>
>> I built unimrcp last night, it was quite straight forward. In theory, if I
>> weren't old and my C/autoconf skills rather atrophied, it wouldn't seem like
>> it would be that huge a deal to port/fix openmrcp to unimrcp.
>>
>> Finally, Anthony I was looking at the Lumenvox path as well, but got
>> deterred by the licensing hassle. This seems to be a universal ASR issue. I
>> would reason I can find the old module in SVN? Were they going to grant
>> "community dev" licenses? Again - I am willing to volunteer to do some
>> testing/doc at least.
>>
>> Andy
>>
>>
>>
>> On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote:
>>
>> If you can get it to break on linux I will ssh in and fix it for you.
>> If you cannot, i can try to fix it for you over rdp but that won't be very
>> fun.
>>
>> We can think about reinstating mod_lumenvox as well as another windows
>> based asr
>> alternative.  I deleted it for the same reason we will probably delete
>> mod_openmrcp because
>> nobody was using it and there was no way to support it because our dev
>> licenses had expired.
>>
>> Lumenvox has offered us some new dev licenses to bring it back but I would
>> need someone to actually want it to work to put in charge of it.
>>
>> We will be clear about what is supported and what is not in the 1.0.2
>> release scheduled
>> to be released in the near future.
>>
>>
>>
>>
>> On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:
>>
>>>  Hi Anthony,
>>>
>>> mod_openmrcp was a contribution to the community by a 3rd party
>>> individual.
>>>
>>> As i have clearly stated in 2 previous emails, the man has decided to
>>> discontinue the openmrcp project.
>>> So now we are left with the remains of the module and discontinued code.
>>> This was not our decision it was his.
>>>
>>>  I absolutely understand this but it's important, from a user point of
>>> view, to be able to know which bits of FS are current/supported and which
>>> aren't.
>>>
>>> Some people use it without issue which may mean that the crash you
>>> reported is windows specific and I do not have a working lab of any mrcp
>>> capbable system to try it against in unix for that matter.  I have a list of
>>> work to do from here to the moon and back so on an issue like this, unless
>>> someone can hand me login credentials to some box and give me a phone number
>>> to dial to reporduce the issue, it will be a long time until we can deal
>>> with it.
>>>
>>>  It's useful to know that there are people using mod_openmrcp without
>>> issue: I did ask here if anyone was a while back, and no-one fessed up.
>>> I'll give it a go on a Linux box and report back.  And if you'd like a
>>> dev/test environment set up, then just tell me which one.
>>>
>>> And the question arises, should we bother working on it anymore if the
>>> lib has been abandoned and we cannot even get any support from it's author
>>> which is where the problem most likely lies.
>>>
>>> I try not to get too annoyed by these remarks about what we *ought to do*
>>> because I know people lose sight of how much of the work to support the
>>> project is done by a small group of 3 people and not the 2000 people it
>>> appears to be from the outside looking in. (I've been answering email for 4
>>> hours now)
>>>
>>>  Those guys who claim to have all that money in an offshore bank account
>>> are lying - you don't have to reply to them in future ;-)  Seriously,
>>> though, I don't think it's too outrageous an idea to document what's
>>> supported and were you (for example) to have suggested that I get in touch
>>> with the contributors to the various modules, ask them what their view of
>>> its status is, condense the answers in to a list and report back, it's
>>> something I'd quite happily do.
>>>
>>> My suggestion is to pool some cash and pay the guy to make mod_unimrcp
>>> for FS that we can maintain in tree knowing the development can be supported
>>> by the original author.
>>>
>>> Quite happy to participate in that, too.. the problem is that I've a demo
>>> to do like yesterday and the timescale for mod_unimrcp is a bit on the long
>>> side for that.  I'd rather not have to do it with Asterisk and Lumenvox..!
>>>
>>> Cheers --
>>>
>>> Dave
>>>
>>>
>>>
>>> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>>>
>>>>  Hi Mike,
>>>>
>>>> My experience is that it's somewhat broken - it took two trivial tweaks
>>>> to get it to work with IBM's ASR and TTS, but there's a more intractable
>>>> problem to do with memory getting overwritten (I assume that this is
>>>> something to do with something being freed when it shouldn't be) which
>>>> causes a segfault on the second or third session after the module being
>>>> loaded.
>>>>
>>>> Without wishing to sound like a stuck record, one thing that you guys
>>>> really ought to do is to decide what's supported and what isn't, and make
>>>> this obvious - for example, move unsupported modules to a different place in
>>>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>>>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>>>> round in circles a bit trying to make it work, and then find out (a) that it
>>>> doesn't and (b) it's not going to be fixed because it's not supported.
>>>>
>>>> Cheers --
>>>>
>>>> Dave
>>>>
>>>>  I would not say it is totally broken, it is known to work in quite a
>>>> few places, but we are unlikely to be doing any new fixes in it.
>>>>  Mike
>>>>
>>>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>>>
>>>>  Hi Anthony,
>>>>
>>>> Oh! OK.
>>>>
>>>> So is this module "totally broken".
>>>>
>>>> I say this because I can't seem to get it to work at all with the
>>>> example in that Mod_openmrcp wiki page but I thought it might because I'm
>>>> not be using the right Cepstral software (freetrial download versus the
>>>> paided for SDK) or that I'm not using the right port numbers or something
>>>> else I didn't do. I used TcpView to look at local port associated with my
>>>> Cepstral software and changed a few things but still nothing. I changed the
>>>> loglevel setting to 7 in the wiki's example but I don't see the kind of
>>>> output on the console that I would expect for debug mode.
>>>>
>>>> Thanks. Mark.
>>>>
>>>>
>>>>    ------------------------------
>>>> _______________________________________________
>>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> --
>>>> David Knell, Director, 3C Limited
>>>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> ------------------------------
>>> _______________________________________________
>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> David Knell, Director, 3C Limited
>>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From sergey.kirillov at gmail.com  Tue Dec  2 10:06:15 2008
From: sergey.kirillov at gmail.com (Sergey Kirillov)
Date: Tue, 02 Dec 2008 20:06:15 +0200
Subject: [Freeswitch-users] Support for Junghanns duoBRI
Message-ID: <49357917.30804@gmail.com>

Cool. Thanks for the answer.

> All HFC-based cards supported by bristuffed Zaptel should work.
>
> Stefan


From kkielhofner at star2star.com  Tue Dec  2 11:03:41 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Tue, 2 Dec 2008 14:03:41 -0500
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
	<191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
Message-ID: <2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>

On 12/2/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Naturally, either way is stupid.

  Word.

> The whole idea of putting the transport in a uri param is equally stupid to
> using 2 different protocol names but since SIP is the descendant of http it
> they decided to stick with the stupidity of http/https and have sip/sips
> which is almost as if it was designed to break all software trying to keep
> up with url syntax.

  Too late now.

> If they are going to insist on using text params you'd think something like
> transport=foo;security=tls would be even *more* flexable in case alternate
> methods to encrypt crop up.

  I can agree with you here...

  URI parameters in SIP have come to be the catch all for random junk
that doesn't seem to fit anywhere else.  Note that "random junk"
includes everything from transport, to number portability, to CICs, to
ISUP-OLI and on.

  Even in my world setting up proxies, UAs, etc to parse out the
various crap people put in SIP URI params is a hassle.  A big one.

  What a mess!!!

> This is, of course, the first step into a lengthy 12 hour discussion on how
> stupid SIP and url/text based
>  protocols are.

  I like them but I'm weird.

> I dare someone to crank up the pcap on a box doing SIP presence for 20
> phones and "read"
> the 1200 byte messages with all kinds of hyeroglyphic url syntax and
> embedded xml payloads and write
>  up a paper on how much "sense" it makes to have it be "readable".

  I do it all the time.  I think it's quite usable.  ngrep provides a
small enough binary and the ability to match on text.  Certainly
easier to use, especially on embedded systems without the luxury of
dedicated protocol decoders.  With a simple ngrep binary I can debug
any text based protocol I understand.

  Of course, turn on TLS and see how useful *any* of these tools are...

  The core SIP spec and authors can't be blamed for the various junk
people have been putting in SIP bodies.  If what's going on in the
real world is any indication, that ship sailed long ago.  At this
point as long as implementations can at least handle multi-part
sensibly and everyone specifies the correct MIME type I don't really
care.

  Even nastier examples abound - embedded, encapsulated ISUP!  How
about GTD?  What about Linksys phones using SIP INFO to serve
directories?  Man I could go on and on...

  I'm not going to write a paper about it but I don't think it's that
bad.  Maybe I'm not just weird; maybe I'm a masochist! :)

> PS
>
> supposedly sofia can support sctp,
> someone should try it.

  That would be cool.  For anyone wanting to try, various SERs support
SCTP.  Cisco gateways do too.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From c_cav_01 at yahoo.com  Tue Dec  2 11:08:13 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 11:08:13 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>
References: <20791453.post@talk.nabble.com>
	<B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>
Message-ID: <20798791.post@talk.nabble.com>


-- filename "dialplan/extensions/13033253678.xml" --   This is the primary
DID assigned.

  <extension name="3033253678">
    <!--<condition field="destination_number"
expression="^[0,1]?3033253678$"/>-->
    <condition field="${sip_to_user}" expression="^([0,1]?)(3033253678)$">
      <action application="ring_ready"/>
      <action application="set" data="${dialed_user}=$2" />
      <action application="set" data="ignore_early_media=true"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESINATION,NO_ANSWER,DESTINATION_OUT_OF_ORDER,USER_BUSY"/>
      <action application="playback"
data="/usr/local/freeswitch/sounds/holdmusic/06.mp3"/>
      <action application="set" data="call_timeout=25"/>
      <action application="bridge" data="user/$2@$${domain}"/>
      <action application="answer"/>
      <action application="voicemail" data="default $${domain} $2"/>
    </condition>
  </extension>
  

-- filename "dialplan/extensions/18162565804.xml" --   This is the primary
DID assigned.

<extension name="8162565804">
    <!--<condition field="destination_number"
expression="^[0,1]?3033253678$"/>-->
    <condition field="${sip_to_user}" expression="^([0,1]?)(8162565804)$">
      <action application="ring_ready"/>
      <action application="set" data="ignore_early_media=true"/>
      <action application="set" data="call_timeout=25"/>
      <action application="set" data="${dialed_user}=$2"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESINATION,NO_ANSWER,DESTINATION_OUT_OF_ORDER"/>
      <action application="playback"
data="/usr/local/freeswitch/sounds/holdmusic/07.mp3"/>
      <action application="bridge" data="user/$2@$${domain}"/>
      <action application="answer"/>
      <action application="voicemail" data="default $${domain} $2"/>
    </condition>
  </extension>

-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20798791.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Tue Dec  2 11:13:46 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 11:13:46 -0800
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
	<191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
	<2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
Message-ID: <87f2f3b90812021113x7a7c5c51v11e80a6ef82c012d@mail.gmail.com>

Bring on SNAP, baby!

On Tue, Dec 2, 2008 at 11:03 AM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On 12/2/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> > Naturally, either way is stupid.
>
>  Word.
>
> > The whole idea of putting the transport in a uri param is equally stupid
> to
> > using 2 different protocol names but since SIP is the descendant of http
> it
> > they decided to stick with the stupidity of http/https and have sip/sips
> > which is almost as if it was designed to break all software trying to
> keep
> > up with url syntax.
>
>  Too late now.
>
> > If they are going to insist on using text params you'd think something
> like
> > transport=foo;security=tls would be even *more* flexable in case
> alternate
> > methods to encrypt crop up.
>
>  I can agree with you here...
>
>  URI parameters in SIP have come to be the catch all for random junk
> that doesn't seem to fit anywhere else.  Note that "random junk"
> includes everything from transport, to number portability, to CICs, to
> ISUP-OLI and on.
>
>  Even in my world setting up proxies, UAs, etc to parse out the
> various crap people put in SIP URI params is a hassle.  A big one.
>
>  What a mess!!!
>
> > This is, of course, the first step into a lengthy 12 hour discussion on
> how
> > stupid SIP and url/text based
> >  protocols are.
>
>  I like them but I'm weird.
>
> > I dare someone to crank up the pcap on a box doing SIP presence for 20
> > phones and "read"
> > the 1200 byte messages with all kinds of hyeroglyphic url syntax and
> > embedded xml payloads and write
> >  up a paper on how much "sense" it makes to have it be "readable".
>
>  I do it all the time.  I think it's quite usable.  ngrep provides a
> small enough binary and the ability to match on text.  Certainly
> easier to use, especially on embedded systems without the luxury of
> dedicated protocol decoders.  With a simple ngrep binary I can debug
> any text based protocol I understand.
>
>  Of course, turn on TLS and see how useful *any* of these tools are...
>
>  The core SIP spec and authors can't be blamed for the various junk
> people have been putting in SIP bodies.  If what's going on in the
> real world is any indication, that ship sailed long ago.  At this
> point as long as implementations can at least handle multi-part
> sensibly and everyone specifies the correct MIME type I don't really
> care.
>
>  Even nastier examples abound - embedded, encapsulated ISUP!  How
> about GTD?  What about Linksys phones using SIP INFO to serve
> directories?  Man I could go on and on...
>
>  I'm not going to write a paper about it but I don't think it's that
> bad.  Maybe I'm not just weird; maybe I'm a masochist! :)
>
> > PS
> >
> > supposedly sofia can support sctp,
> > someone should try it.
>
>  That would be cool.  For anyone wanting to try, various SERs support
> SCTP.  Cisco gateways do too.
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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From anthony.minessale at gmail.com  Tue Dec  2 11:16:36 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 13:16:36 -0600
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
	<191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
	<2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
Message-ID: <191c3a030812021116n724f77c8oe477a1585f12e8da@mail.gmail.com>

We'll schedule a round table with the topic

SIP OMFG STFU

At the next ClueCon aug 4th-6th 2009 to stir things up a bit =D


On Tue, Dec 2, 2008 at 1:03 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On 12/2/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> > Naturally, either way is stupid.
>
>   Word.
>
> > The whole idea of putting the transport in a uri param is equally stupid
> to
> > using 2 different protocol names but since SIP is the descendant of http
> it
> > they decided to stick with the stupidity of http/https and have sip/sips
> > which is almost as if it was designed to break all software trying to
> keep
> > up with url syntax.
>
>   Too late now.
>
> > If they are going to insist on using text params you'd think something
> like
> > transport=foo;security=tls would be even *more* flexable in case
> alternate
> > methods to encrypt crop up.
>
>   I can agree with you here...
>
>  URI parameters in SIP have come to be the catch all for random junk
> that doesn't seem to fit anywhere else.  Note that "random junk"
> includes everything from transport, to number portability, to CICs, to
> ISUP-OLI and on.
>
>  Even in my world setting up proxies, UAs, etc to parse out the
> various crap people put in SIP URI params is a hassle.  A big one.
>
>  What a mess!!!
>
> > This is, of course, the first step into a lengthy 12 hour discussion on
> how
> > stupid SIP and url/text based
> >  protocols are.
>
>   I like them but I'm weird.
>
> > I dare someone to crank up the pcap on a box doing SIP presence for 20
> > phones and "read"
> > the 1200 byte messages with all kinds of hyeroglyphic url syntax and
> > embedded xml payloads and write
> >  up a paper on how much "sense" it makes to have it be "readable".
>
>   I do it all the time.  I think it's quite usable.  ngrep provides a
> small enough binary and the ability to match on text.  Certainly
> easier to use, especially on embedded systems without the luxury of
> dedicated protocol decoders.  With a simple ngrep binary I can debug
> any text based protocol I understand.
>
>  Of course, turn on TLS and see how useful *any* of these tools are...
>
>  The core SIP spec and authors can't be blamed for the various junk
> people have been putting in SIP bodies.  If what's going on in the
> real world is any indication, that ship sailed long ago.  At this
> point as long as implementations can at least handle multi-part
> sensibly and everyone specifies the correct MIME type I don't really
> care.
>
>  Even nastier examples abound - embedded, encapsulated ISUP!  How
> about GTD?  What about Linksys phones using SIP INFO to serve
> directories?  Man I could go on and on...
>
>  I'm not going to write a paper about it but I don't think it's that
> bad.  Maybe I'm not just weird; maybe I'm a masochist! :)
>
> > PS
> >
> > supposedly sofia can support sctp,
> > someone should try it.
>
>   That would be cool.  For anyone wanting to try, various SERs support
> SCTP.  Cisco gateways do too.
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20081202/bea4d08e/attachment-0001.html 

From kkielhofner at star2star.com  Tue Dec  2 11:20:31 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Tue, 2 Dec 2008 14:20:31 -0500
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <191c3a030812021116n724f77c8oe477a1585f12e8da@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
	<191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
	<2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
	<191c3a030812021116n724f77c8oe477a1585f12e8da@mail.gmail.com>
Message-ID: <2d9149cd0812021120m2774d712qe95c9c32ecdfb85b@mail.gmail.com>

On 12/2/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> We'll schedule a round table with the topic
>
> SIP OMFG STFU
>
> At the next ClueCon aug 4th-6th 2009 to stir things up a bit =D
>

Heh.  I've been trying to make it back these last couple of years.  I
just might make it in '09!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From c_cav_01 at yahoo.com  Tue Dec  2 11:29:03 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 11:29:03 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20798791.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com>
	<B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>
	<20798791.post@talk.nabble.com>
Message-ID: <20799146.post@talk.nabble.com>


Note: while reading up on regex, I see that the ',' in ([0,1]) is superflous,
has been removed.  regex is now:
^([01]?)(8162565804)$
Didn't fix the problem but I'm a perfectionist, had to be changed. :D
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20799146.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From per_moeller at mac.com  Tue Dec  2 11:31:59 2008
From: per_moeller at mac.com (=?iso-8859-1?Q?Per_M=F8ller?=)
Date: Tue, 02 Dec 2008 20:31:59 +0100
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
References: <000001c9530d$912d86d0$b3889470$@com>
	<5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
Message-ID: <000f01c954b4$a616fa60$f244ef20$@com>

I checked out the trunk version, and it's still slow. However I found one
improvement - it does not crash on shutdown anymore.

Could anymore give me some pointers on how to try to debug this on the
Windows platform?


// Per

Fra: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Carlos
Talbot
Sendt: 2. december 2008 17:13
Til: freeswitch-users at lists.freeswitch.org
Emne: Re: [Freeswitch-users] Windows is slow?

Have you tried the latest msi build? It's based off svn 10564.

Carlos

On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:
I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
on a Windows Vista machine using the precompiled .msi - actually the same
machine).

Using the default configuration files, and using 2 Snom 360 phones I dialed
from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
instantly, but under Windows it takes 1-2 seconds before it starts ringing.

It seems to be in the dialplan the time is spent. From the time I see this
line on the console:

[INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
context default

Until the next thing happens it always takes at least 1 full second, but on
the Mac it happens instantly.

Why is the Windows build this much slower? Is it a known problem?

I get the feeling that the majority of the FS community is Unix based, which
is fine by me, but I would really like to know just how well supported and
stable the Win32 build is and if this is currently a viable way to go, or if
I should stick to Linux/BSD/Mac for production use?


// Per


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From mgg at giagnocavo.net  Tue Dec  2 11:33:48 2008
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 2 Dec 2008 14:33:48 -0500
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <000f01c954b4$a616fa60$f244ef20$@com>
References: <000001c9530d$912d86d0$b3889470$@com>
	<5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
	<000f01c954b4$a616fa60$f244ef20$@com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702336DC03C@mse17be1.mse17.exchange.ms>

Can you do a console loglevel debug, then send all the output around that time?

Apart from that, the quickest way might just to attach a debugger, then break all when it pauses and see where the threads are :).

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Per M?ller
Sent: Tuesday, December 02, 2008 12:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Windows is slow?

I checked out the trunk version, and it's still slow. However I found one
improvement - it does not crash on shutdown anymore.

Could anymore give me some pointers on how to try to debug this on the
Windows platform?


// Per

Fra: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Carlos
Talbot
Sendt: 2. december 2008 17:13
Til: freeswitch-users at lists.freeswitch.org
Emne: Re: [Freeswitch-users] Windows is slow?

Have you tried the latest msi build? It's based off svn 10564.

Carlos

On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:
I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
on a Windows Vista machine using the precompiled .msi - actually the same
machine).

Using the default configuration files, and using 2 Snom 360 phones I dialed
from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
instantly, but under Windows it takes 1-2 seconds before it starts ringing.

It seems to be in the dialplan the time is spent. From the time I see this
line on the console:

[INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
context default

Until the next thing happens it always takes at least 1 full second, but on
the Mac it happens instantly.

Why is the Windows build this much slower? Is it a known problem?

I get the feeling that the majority of the FS community is Unix based, which
is fine by me, but I would really like to know just how well supported and
stable the Win32 build is and if this is currently a viable way to go, or if
I should stick to Linux/BSD/Mac for production use?


// Per


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From mrjoebain at gmail.com  Tue Dec  2 06:29:12 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 2 Dec 2008 14:29:12 +0000
Subject: [Freeswitch-users] Javascript ODBC on Windows
Message-ID: <748d53500812020629p6a0d178dh672cec871c018254@mail.gmail.com>

Hi all,

Is it possible to use mod_spidermonkey_odbc with a Windows installation of
FreeSWITCH at the moment? If so does anyone have any pointers? I get:

2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145 switch_odbc_handle_connect()
Connecting ivr_test
2008-12-02 14:23:57 [ERR] switch_odbc.c:160 switch_odbc_handle_connect()
STATE: IM002 CODE 0 ERROR: [Microsoft][ODBC Driver Manager] Data source name
not found and no default driver specified

when I try.

Thanks in advance,

Joe Bain
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From anthony.minessale at gmail.com  Tue Dec  2 12:05:15 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 14:05:15 -0600
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702336DC03C@mse17be1.mse17.exchange.ms>
References: <000001c9530d$912d86d0$b3889470$@com>
	<5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
	<000f01c954b4$a616fa60$f244ef20$@com>
	<6E8D2069C08AA84A83D336E996AE4C6702336DC03C@mse17be1.mse17.exchange.ms>
Message-ID: <191c3a030812021205r619ad735le129731ccb8f69d0@mail.gmail.com>

is it stun timeout ?

do you have one of the ip set to stun:foo ?


On Tue, Dec 2, 2008 at 1:33 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

> Can you do a console loglevel debug, then send all the output around that
> time?
>
> Apart from that, the quickest way might just to attach a debugger, then
> break all when it pauses and see where the threads are :).
>
> -Michael
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Per M?ller
> Sent: Tuesday, December 02, 2008 12:32 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Windows is slow?
>
> I checked out the trunk version, and it's still slow. However I found one
> improvement - it does not crash on shutdown anymore.
>
> Could anymore give me some pointers on how to try to debug this on the
> Windows platform?
>
>
> // Per
>
> Fra: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Carlos
> Talbot
> Sendt: 2. december 2008 17:13
> Til: freeswitch-users at lists.freeswitch.org
> Emne: Re: [Freeswitch-users] Windows is slow?
>
> Have you tried the latest msi build? It's based off svn 10564.
>
> Carlos
>
> On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:
> I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
> on a Windows Vista machine using the precompiled .msi - actually the same
> machine).
>
> Using the default configuration files, and using 2 Snom 360 phones I dialed
> from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
> instantly, but under Windows it takes 1-2 seconds before it starts ringing.
>
> It seems to be in the dialplan the time is spent. From the time I see this
> line on the console:
>
> [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
> context default
>
> Until the next thing happens it always takes at least 1 full second, but on
> the Mac it happens instantly.
>
> Why is the Windows build this much slower? Is it a known problem?
>
> I get the feeling that the majority of the FS community is Unix based,
> which
> is fine by me, but I would really like to know just how well supported and
> stable the Win32 build is and if this is currently a viable way to go, or
> if
> I should stick to Linux/BSD/Mac for production use?
>
>
> // Per
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Tue Dec  2 12:32:49 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 12:32:49 -0800
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
Message-ID: <87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>

Right now this page is up-to-date with the latest info:
http://wiki.freeswitch.org/wiki/Mod_fax

T.38 is not (yet) supported.

-MC

On Tue, Dec 2, 2008 at 9:40 AM, Dennis <odermann at googlemail.com> wrote:

> hi,
>
> because we do not get tired of testing and playing a lot with the
> beloved fs, we now arrived at the fax feature :-)
>
> i am not sure if the docs are up to date or if there was a lot of
> development in the meantime. therefore i would like to ask, what is
> possible and what will come in the near future.
>
> we are using fs, socket outbound and php and would like to make
> something like fax to mail as an additional service.
>
> is t38 supported?
> can i pass incoming faxes over the same socket as calls?
> can i convert faxes into pdf?
> is fax over sip reliable (as far as i have heard, under asterisk fax
> is nothing one should use)?
> and so on, and so on....
>
> i would be very happy to hear some user experiences with fs and fax.
> if it seems, that we can use fax with over socket outbound, we will do
> hardcore testing ;-)
>
> thanks,
> dennis
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kkielhofner at star2star.com  Tue Dec  2 13:28:28 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Tue, 2 Dec 2008 16:28:28 -0500
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
Message-ID: <2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>

On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <msc at freeswitch.org> wrote:
> Right now this page is up-to-date with the latest info:
> http://wiki.freeswitch.org/wiki/Mod_fax
>
> T.38 is not (yet) supported.
>
> -MC
>

Can you (or someone) elaborate on this?  Maybe the answer really is
no, but what about support for UDPTL, pass through, etc?

It looks like Sofia should be good to go...

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From mike at jerris.com  Tue Dec  2 14:36:34 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 17:36:34 -0500
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
Message-ID: <40890CF2-2279-464A-A58D-A86087D2CD1A@jerris.com>

T.38 passthrough IS supported, T.38 endpoint and gateway are not yet  
supported.

Mike

On Dec 2, 2008, at 4:28 PM, Kristian Kielhofner wrote:

> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <msc at freeswitch.org>  
> wrote:
>> Right now this page is up-to-date with the latest info:
>> http://wiki.freeswitch.org/wiki/Mod_fax
>>
>> T.38 is not (yet) supported.
>>
>> -MC
>>
>
> Can you (or someone) elaborate on this?  Maybe the answer really is
> no, but what about support for UDPTL, pass through, etc?
>
> It looks like Sofia should be good to go...
>


From mike at jerris.com  Tue Dec  2 14:38:40 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 17:38:40 -0500
Subject: [Freeswitch-users] Javascript ODBC on Windows
In-Reply-To: <748d53500812020629p6a0d178dh672cec871c018254@mail.gmail.com>
References: <748d53500812020629p6a0d178dh672cec871c018254@mail.gmail.com>
Message-ID: <E05E9383-0BF2-4B8B-93A5-09FB99817F19@jerris.com>

Yes, it should work fine.  As the error message says it didn't find  
the data source name you specified.  You need to setup your odbc data  
source on the system

Mike

On Dec 2, 2008, at 9:29 AM, Joe Bain wrote:

> Hi all,
>
> Is it possible to use mod_spidermonkey_odbc with a Windows  
> installation of FreeSWITCH at the moment? If so does anyone have any  
> pointers? I get:
>
> 2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145  
> switch_odbc_handle_connect() Connecting ivr_test
> 2008-12-02 14:23:57 [ERR] switch_odbc.c:160  
> switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [Microsoft] 
> [ODBC Driver Manager] Data source name not found and no default  
> driver specified
>
> when I try.
>
> Thanks in advance,
>
> Joe Bain
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Dec  2 14:39:06 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 16:39:06 -0600
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20799146.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com>
	<B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>
	<20798791.post@talk.nabble.com> <20799146.post@talk.nabble.com>
Message-ID: <AF0CB43E-C3D1-442B-A29F-0DBE21E71025@freeswitch.org>

After you set ${dialed_user}=$2 try using ${dialed_user} everywhere  
instead of $2 just to test.

/b

On Dec 2, 2008, at 1:29 PM, ccav wrote:

>
> Note: while reading up on regex, I see that the ',' in ([0,1]) is  
> superflous,
> has been removed.  regex is now:
> ^([01]?)(8162565804)$
> Didn't fix the problem but I'm a perfectionist, had to be changed. :D
> -- 
> View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20799146.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From c_cav_01 at yahoo.com  Tue Dec  2 16:05:47 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 16:05:47 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20791453.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com>
Message-ID: <20803931.post@talk.nabble.com>


Made the change, no joy.

Do I need to set sip_req_user to the updated DID?

Also, I misspoke in my first post, apparently the bridge is NOT going
through either.  Is there some var/param I can set with $2 so I can see it
in the "info"?
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20803931.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From c_cav_01 at yahoo.com  Tue Dec  2 16:33:38 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 16:33:38 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20803931.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com> <20803931.post@talk.nabble.com>
Message-ID: <20804247.post@talk.nabble.com>


Okay, I found out who the culprit is, but I still want to find a fix so the
dialplan works like I want.

The <application="playback" causes an answer and early media...  When I
comment that out, the voicemail works almost properly.  It's clipping the
(81) off the front of the 816 DID, which is probably thinking those are pbx
dialout values.  If anyone can tell me where that is set so I can change
it...  But the bridge and voicemail both work like they are sposed to.

However, something is happening in playback that is making it impossible to
bridge the call or send it to voicemail properly.

-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804247.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From c_cav_01 at yahoo.com  Tue Dec  2 16:35:59 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 16:35:59 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
Message-ID: <20804247.post@talk.nabble.com>


Okay, I found out who the culprit is, but I still want to find a fix so the
dialplan works like I want.

The <application="playback" causes an answer and early media...  When I
comment that out, the voicemail works almost properly.  It's clipping the
(81) off the front of the 816 DID, which is probably thinking those are pbx
dialout values.  If anyone can tell me where that is set so I can change
it...  But the bridge and voicemail both work like they are sposed to.

However, something is happening in playback that is making it impossible to
bridge the call or send it to voicemail properly.

Instead of calling application playback and then trying to bridge, should I
just bridge to both?  What I want is the caller to hear my music instead of
the ring, and maybe this is the wrong way to do it?
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804247.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Tue Dec  2 16:48:10 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 16:48:10 -0800
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
Message-ID: <87f2f3b90812021648v7f1402ddma747ea0da1eac577@mail.gmail.com>

On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Right now this page is up-to-date with the latest info:
> > http://wiki.freeswitch.org/wiki/Mod_fax
> >
> > T.38 is not (yet) supported.
> >
> > -MC
> >
>
> Can you (or someone) elaborate on this?  Maybe the answer really is
> no, but what about support for UDPTL, pass through, etc?
>

Excellent questions! I will research and report back to the list...
-MC


>
> It looks like Sofia should be good to go...
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From c_cav_01 at yahoo.com  Tue Dec  2 17:11:52 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 17:11:52 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20804247.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com> <20803931.post@talk.nabble.com>
	<20804247.post@talk.nabble.com>
Message-ID: <20804652.post@talk.nabble.com>


RESOLVED.

Duh, I'm sposed to use ringback, not playback...

Someone should write a book on this...  Maybe I will.
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804652.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Tue Dec  2 17:18:22 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 17:18:22 -0800
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
Message-ID: <87f2f3b90812021718j5aae69aav6dd8ee7953e2b1ff@mail.gmail.com>

Kristian,

Are you on the IRC channel by any chance?
-MC (IRC: mercutioviz)

On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Right now this page is up-to-date with the latest info:
> > http://wiki.freeswitch.org/wiki/Mod_fax
> >
> > T.38 is not (yet) supported.
> >
> > -MC
> >
>
> Can you (or someone) elaborate on this?  Maybe the answer really is
> no, but what about support for UDPTL, pass through, etc?
>
> It looks like Sofia should be good to go...
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Dec  2 17:18:53 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 17:18:53 -0800
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20804652.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com> <20803931.post@talk.nabble.com>
	<20804247.post@talk.nabble.com> <20804652.post@talk.nabble.com>
Message-ID: <87f2f3b90812021718k1b0c51dcx8740e3c387ce4887@mail.gmail.com>

hehe, careful what you wish for...

On Tue, Dec 2, 2008 at 5:11 PM, ccav <c_cav_01 at yahoo.com> wrote:

>
> RESOLVED.
>
> Duh, I'm sposed to use ringback, not playback...
>
> Someone should write a book on this...  Maybe I will.
> --
> View this message in context:
> http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804652.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From klaus.teller at gmx.net  Tue Dec  2 19:15:52 2008
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 03 Dec 2008 04:15:52 +0100
Subject: [Freeswitch-users] Bridging from Event Socket API
Message-ID: <20081203031552.178560@gmx.net>

Hi Folks,

so far i could understand how to bridge calls with Javascript. I'm trying to do the same with Java via the Socket Interface. My first trials weren't successful. maybe you can help me understand what is goin on. 

What i want to do is to bridge an existing leg (Unique-ID is known) to a party that wasn't yet dialed (Unique-ID unknown). With javascript it is something like:

session.bridge("sofia/internal/1002");

How do i do this using the event socket interface? what application/command would i use with which arguments?


One way i tried to do this is to orginate a call to 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't successful. The only message i had on the FS console is: 

2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693 switch_core_session_queue_private_event() Send signal sofia/internal/1001 at 192.168.1.121 [BREAK]

Any idea what i'm missing?

Thanks,

Klaus.





-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From msc at freeswitch.org  Tue Dec  2 20:39:08 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 20:39:08 -0800
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <20081203031552.178560@gmx.net>
References: <20081203031552.178560@gmx.net>
Message-ID: <87f2f3b90812022039odcacf9dte67c33707e41efc0@mail.gmail.com>

You probably have several options depending upon your needs. Could you
elaborate a bit on what the big picture is? Also, what exactly were you
doing when you established the second call leg? Did the second call let get
created and a valid uuid assigned, etc.? Just checking.

Let us know,
MC

On Tue, Dec 2, 2008 at 7:15 PM, Klaus Teller <klaus.teller at gmx.net> wrote:

> Hi Folks,
>
> so far i could understand how to bridge calls with Javascript. I'm trying
> to do the same with Java via the Socket Interface. My first trials weren't
> successful. maybe you can help me understand what is goin on.
>
> What i want to do is to bridge an existing leg (Unique-ID is known) to a
> party that wasn't yet dialed (Unique-ID unknown). With javascript it is
> something like:
>
> session.bridge("sofia/internal/1002");
>
> How do i do this using the event socket interface? what application/command
> would i use with which arguments?
>
>
> One way i tried to do this is to orginate a call to 'sofia/internal/1002'
> and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't
> successful. The only message i had on the FS console is:
>
> 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693
> switch_core_session_queue_private_event() Send signal sofia/internal/
> 1001 at 192.168.1.121 [BREAK]
>
> Any idea what i'm missing?
>
> Thanks,
>
> Klaus.
>
>
>
>
>
> --
> Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen:
> http://www.gmx.net/de/go/multimessenger
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dave at 3c.co.uk  Wed Dec  3 03:15:57 2008
From: dave at 3c.co.uk (David Knell)
Date: Wed, 03 Dec 2008 11:15:57 +0000
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <20081203031552.178560@gmx.net>
References: <20081203031552.178560@gmx.net>
Message-ID: <49366A6D.4000702@3c.co.uk>

Hi Klaus,

Some Perl code snippets - we use:
call_command("bridge", "sofia/gateway/bt/$ntd");
which, in turn, is:
sub call_command($$) {
        my $cmd = shift;
        my $arg = shift;
        print $sock "sendmsg\ncall-command: execute\nexecute-app-name: 
$cmd\nexecute-app-arg: $arg\n\n";
}

Cheers --

Dave

> Hi Folks,
>
> so far i could understand how to bridge calls with Javascript. I'm trying to do the same with Java via the Socket Interface. My first trials weren't successful. maybe you can help me understand what is goin on. 
>
> What i want to do is to bridge an existing leg (Unique-ID is known) to a party that wasn't yet dialed (Unique-ID unknown). With javascript it is something like:
>
> session.bridge("sofia/internal/1002");
>
> How do i do this using the event socket interface? what application/command would i use with which arguments?
>
>
> One way i tried to do this is to orginate a call to 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't successful. The only message i had on the FS console is: 
>
> 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693 switch_core_session_queue_private_event() Send signal sofia/internal/1001 at 192.168.1.121 [BREAK]
>
> Any idea what i'm missing?
>
> Thanks,
>
> Klaus.
>
>
>
>
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 



From regs at kinetix.gr  Wed Dec  3 05:30:00 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 15:30:00 +0200
Subject: [Freeswitch-users] How to get info from the b-leg
Message-ID: <493689D8.9040708@kinetix.gr>

Hi,

    I am making a simple bridge between two call legs :

Client --(a-leg)--> FS --(b-leg)-->Provider

How can I get information like network-address of the Provider, 
media-address,
port used, media port used etc. from the second leg (b-leg)?

Is all the information provided by the a-leg available for the b-leg as 
well? If, yes
how can I access it? (and log it to my CDR file eventually)


From anthony.minessale at gmail.com  Wed Dec  3 05:53:22 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 07:53:22 -0600
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <493689D8.9040708@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
Message-ID: <191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>

2 options.
1) enable b-leg logging on the cdr module.
2) you can use the prefix bleg_ in a variable context to get to
caller_profile members
from the b leg.

eg ${bleg_caller_id_name}


On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:

> Hi,
>
>    I am making a simple bridge between two call legs :
>
> Client --(a-leg)--> FS --(b-leg)-->Provider
>
> How can I get information like network-address of the Provider,
> media-address,
> port used, media port used etc. from the second leg (b-leg)?
>
> Is all the information provided by the a-leg available for the b-leg as
> well? If, yese
> how can I access it? (and log it to my CDR file eventually)
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From regs at kinetix.gr  Wed Dec  3 06:18:31 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 16:18:31 +0200
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
Message-ID: <49369537.6040008@kinetix.gr>

b-leg logging is enabled in the cdr module. but in the cdrs I cannot get 
any variables that refer to the b-leg.

I tried the second way using ${sip_to_host} and {bleg_sip_to_host} but :

a) the variable returns the FS IP on the a-leg CDR (correctly)
b) the variable returns nothing on the b-leg CDR (empty). Shouldn't it 
return the "to" host of the b-leg (my providers address)?


Anthony Minessale wrote:
> 2 options.
> 1) enable b-leg logging on the cdr module.
> 2) you can use the prefix bleg_ in a variable context to get to 
> caller_profile members
> from the b leg.
>
> eg ${bleg_caller_id_name}
>
>
> On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr 
> <mailto:regs at kinetix.gr> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     Hi,
>
>        I am making a simple bridge between two call legs :
>
>     Client --(a-leg)--> FS --(b-leg)-->Provider
>
>     How can I get information like network-address of the Provider,
>     media-address,
>     port used, media port used etc. from the second leg (b-leg)?
>
>     Is all the information provided by the a-leg available for the
>     b-leg as
>     well? If, yese
>     how can I access it? (and log it to my CDR file eventually)
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From r.pankratz at fh-wolfenbuettel.de  Wed Dec  3 06:21:16 2008
From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz)
Date: Wed, 03 Dec 2008 15:21:16 +0100
Subject: [Freeswitch-users] Dialing tone when placing a call
	with	portaudio
In-Reply-To: <ADE59C30-1DBE-4DB5-B96B-9EDA0A0DCE07@jerris.com>
References: <20081120154859.16192.qmail@server15.citromail.hu>
	<4934FBED.7030307@fh-wolfenbuettel.de>
	<ADE59C30-1DBE-4DB5-B96B-9EDA0A0DCE07@jerris.com>
Message-ID: <493695DC.4060706@fh-wolfenbuettel.de>

You were right, I was calling sip when not getting a dialing tone.
In the SIP flow I get a 180 ringing and no 183.
Setting ringback channel fixed that issue.

Thanks for your help!

Ren?
> What are you calling, sip I assume, this may be a case where the sip  
> signaling is sending a 180 ringing instead of a 183 and we are not  
> generating ringback in that case.  Can you please confirm that and  
> test if setting the ringback channel variable before bridge fixes this  
> issue?
>
> Mike
>
> On Dec 2, 2008, at 4:12 AM, Rene Pankratz wrote:
>
>   
>> Hello,
>> when using mod_portaudio for calling somebody I don't hear anything
>> until the other party answers the call. Is it possible to play a  
>> dialing
>> tone when the other party is ringing?
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   




From klaus.teller at gmx.net  Wed Dec  3 06:48:02 2008
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 03 Dec 2008 15:48:02 +0100
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <49366A6D.4000702@3c.co.uk>
References: <20081203031552.178560@gmx.net> <49366A6D.4000702@3c.co.uk>
Message-ID: <20081203144802.281260@gmx.net>

Hi All,

Thanks for your feedback. I must be doing something fundamentally wrong. Inbound socket is working without problems. But the exact things that i do on inbound socket, i'm not able to replcate them on outbound socket.

The global picture: I have on Xlite registered at extension 1002 and another one at extension 1003.  Then i have an extension 8998 in the default context. Here is the extension definition:

   <extension name="outbound-test">
        <condition field="destination_number" expression="^8998$" >
            <action application="socket" data="192.168.50.56:10000 full" />
        </condition>		
    </extension>

I use Xlite-1003 to call this extension (8998) and the call is properly notified to the remote Java server. 

Then on the Java side, after receiving the event, i send a CONNECT command: "Connect\n\n"
The answer from Freeswitch is the state of the channel ( a set of variable, value pair).

Up to this point everything seems normal to me. But then, i try to send an answer command:

sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
call-command: execute
execute-app-name: answer
execute-app-arg:

Freswitch replies with: 

Content-Type: command/reply
Reply-Text: +OK

But the call is still not answered. Nothing happens on the freeswitch console (Log level DEBUG) and the dialing XLite is still in calling modus.

Then i try bridging the call to 1002:

sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
call-command: execute
execute-app-name: bridge
execute-app-arg: sofia/internal/1002%192.168.50.94
   
Again Freeswitch does answer with:

Content-Type: command/reply
Reply-Text: +OK

And yet again, nothing is really happening.

What am i missing here?

Thanks,
Klaus.

-------- Original-Nachricht --------
> Datum: Wed, 03 Dec 2008 11:15:57 +0000
> Von: David Knell <dave at 3c.co.uk>
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Bridging from Event Socket API

> Hi Klaus,
> 
> Some Perl code snippets - we use:
> call_command("bridge", "sofia/gateway/bt/$ntd");
> which, in turn, is:
> sub call_command($$) {
>         my $cmd = shift;
>         my $arg = shift;
>         print $sock "sendmsg\ncall-command: execute\nexecute-app-name: 
> $cmd\nexecute-app-arg: $arg\n\n";
> }
> 
> Cheers --
> 
> Dave
> 
> > Hi Folks,
> >
> > so far i could understand how to bridge calls with Javascript. I'm
> trying to do the same with Java via the Socket Interface. My first trials
> weren't successful. maybe you can help me understand what is goin on. 
> >
> > What i want to do is to bridge an existing leg (Unique-ID is known) to a
> party that wasn't yet dialed (Unique-ID unknown). With javascript it is
> something like:
> >
> > session.bridge("sofia/internal/1002");
> >
> > How do i do this using the event socket interface? what
> application/command would i use with which arguments?
> >
> >
> > One way i tried to do this is to orginate a call to
> 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it
> wasn't successful. The only message i had on the FS console is: 
> >
> > 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693
> switch_core_session_queue_private_event() Send signal sofia/internal/1001 at 192.168.1.121
> [BREAK]
> >
> > Any idea what i'm missing?
> >
> > Thanks,
> >
> > Klaus.
> >
> >
> >
> >
> >
> >   
> 
> 
> -- 
> David Knell, Director, 3C Limited
> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
> http://www.3c.co.uk 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From yudha2008 at gmail.com  Wed Dec  3 07:32:12 2008
From: yudha2008 at gmail.com (Baskar)
Date: Wed, 3 Dec 2008 21:02:12 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
Message-ID: <d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>

Hi,

*I have newly installed freeswitch in another machine.

**After starting the freeswitch I try to dial a extension from console but
when i call extension 1002 from freeswitch console, call is connected to
extension 1002, but FS is aborted but call is established in1002.*

*When i dial from console the call get connected and freeswitch is cut.*

*OUtput:*


*FreeSWITCH Version 1.0.trunk (10567) Started.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
2008-12-03 21:02:21 [CONSOLE] mod_local_stream.c:142 read_stream_thread()
Can't open directory: /usr/local/freeswitch/sounds/music/16000*
*
freeswitch at hp30094686650.optimus.co.in> pa devlist*
*API CALL [pa(devlist)] output:
0;/dev/dsp;16;4
1;Intel ICH5: Intel ICH5 (hw:0,0);2;6
2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0
3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0
4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0
5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2
6;front;0;6
7;surround40;0;4
8;surround41;0;128
9;surround50;0;128
10;surround51;0;6
11;iec958;0;2
12;spdif;0;2
13;default;128;128
14;dmix;0;2*

*freeswitch at hp30094686650.optimus.co.in> pa call 1002*
*2008-12-03 21:06:11 [NOTICE] switch_channel.c:564 switch_channel_set_name()
New Channel portaudio/1002 [fae97d5b-3480-410e-af0a-192d00710537]
freeswitch at hp30094686650.optimus.co.in> 2008-12-03 21:06:12 [NOTICE]
mod_portaudio.c:1586 place_call() Channel [portaudio/1002] has been answered
API CALL [pa(call 1002)] output:
SUCCESS:1:fae97d5b-3480-410e-af0a-192d00710537

2008-12-03 21:06:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
FreeSWITCH->1002 in context default
2008-12-03 21:06:12 [WARNING] switch_ivr.c:1840 switch_ivr_set_user() can't
find user [default@]
2008-12-03 21:06:12 [INFO] mod_dptools.c:872 info_function() CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [portaudio/1002]
Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537]
Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [L16]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [L16]
Channel-Write-Codec-Rate: [8000]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [FreeSWITCH]
Caller-Caller-ID-Number: [0000000000]
Caller-Network-Addr: [172.20.176.32]
Caller-Destination-Number: [1002]
Caller-Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537]
Caller-Source: [mod_portaudio]
Caller-Context: [default]
Caller-Channel-Name: [portaudio/1002]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1228318571584600]
Caller-Channel-Created-Time: [1228318571584600]
Caller-Channel-Answered-Time: [1228318572164620]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_channel_name: [portaudio/1002]
variable_endpoint_disposition: [ANSWER]
variable_read_codec: [L16]
variable_read_rate: [8000]
variable_write_codec: [L16]
variable_write_rate: [8000]
variable_use_profile: [nat]
variable_dialed_ext: [1002]
variable_current_application: [info]


2008-12-03 21:06:12 [INFO] mod_dptools.c:858 log_function() Answer-State []n
2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML
features
2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2
record_session::/usr/local/freeswitch/recordings/0000000000.2008-12-03-21-06-12.wav
2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML
features
2008-12-03 21:06:12 [NOTICE] switch_channel.c:564 switch_channel_set_name()
New Channel sofia/internal/sip:1002 at 172.20.179.201:37046;rinstance=e6259d34a17a130a
[e9a905cd-dc7c-49b1-b3f7-1cd52c1129d1]*
*freeswitch: src/switch_core_io.c:202: switch_core_session_read_frame:
Assertion `(*frame)->codec != ((void *)0)' failed.
Aborted (core dumped)
[root at hp30094686650 bin]# *
*

After installing current svn trunk also i get the same error.I cant able to
recover the failure .Correct me were i am wrong.


Thanks Regards,
N.Baskar
*
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From anthony.minessale at gmail.com  Wed Dec  3 07:37:07 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 09:37:07 -0600
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <49369537.6040008@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
	<49369537.6040008@kinetix.gr>
Message-ID: <191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>

outgoing calls to not have an ip value set.
if you want to store the dest ip in the cdr you need to set it as a custom
variable and insert it
into your template for csv cdr or it will just be there in xml cdr

On Wed, Dec 3, 2008 at 8:18 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:

>  b-leg logging is enabled in the cdr module. but in the cdrs I cannot get
> any variables that refer to the b-leg.
>
> I tried the second way using ${sip_to_host} and {bleg_sip_to_host} but :
>
> a) the variable returns the FS IP on the a-leg CDR (correctly)
> b) the variable returns nothing on the b-leg CDR (empty). Shouldn't it
> return the "to" host of the b-leg (my providers address)?
>
>
> Anthony Minessale wrote:
>
> 2 options.
> 1) enable b-leg logging on the cdr module.
> 2) you can use the prefix bleg_ in a variable context to get to
> caller_profile members
> from the b leg.
>
> eg ${bleg_caller_id_name}
>
>
> On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:
>
>> Hi,
>>
>>    I am making a simple bridge between two call legs :
>>
>> Client --(a-leg)--> FS --(b-leg)-->Provider
>>
>> How can I get information like network-address of the Provider,
>> media-address,
>> port used, media port used etc. from the second leg (b-leg)?
>>
>> Is all the information provided by the a-leg available for the b-leg as
>> well? If, yese
>> how can I access it? (and log it to my CDR file eventually)
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From regs at kinetix.gr  Wed Dec  3 07:48:03 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 17:48:03 +0200
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
References: <493689D8.9040708@kinetix.gr>	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>	<49369537.6040008@kinetix.gr>
	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
Message-ID: <4936AA33.7080301@kinetix.gr>

I looked in the b-leg xml cdr and the ip address is not there (for 
signaling) it is only there
for media (${remote_media_ip}) which is not the same thing now, is it?

While we are at it, I noticed that the ${local_media_port} and 
${remote_media_port}
have the same value for each CDR (a or b leg). Shouldn't the first 
variable hold the port
of the FS (on both legs) and the second variable the port of the client 
(in the a-leg) or the port of
the provider (in the b-leg)?

Anthony Minessale wrote:
> outgoing calls to not have an ip value set.
> if you want to store the dest ip in the cdr you need to set it as a 
> custom variable and insert it
> into your template for csv cdr or it will just be there in xml cdr
>
> On Wed, Dec 3, 2008 at 8:18 AM, regs at kinetix.gr 
> <mailto:regs at kinetix.gr> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     b-leg logging is enabled in the cdr module. but in the cdrs I
>     cannot get any variables that refer to the b-leg.
>
>     I tried the second way using ${sip_to_host} and {bleg_sip_to_host}
>     but :
>
>     a) the variable returns the FS IP on the a-leg CDR (correctly)
>     b) the variable returns nothing on the b-leg CDR (empty).
>     Shouldn't it return the "to" host of the b-leg (my providers address)?
>
>
>     Anthony Minessale wrote:
>>     2 options.
>>     1) enable b-leg logging on the cdr module.
>>     2) you can use the prefix bleg_ in a variable context to get to
>>     caller_profile members
>>     from the b leg.
>>
>>     eg ${bleg_caller_id_name}
>>
>>
>>     On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr
>>     <mailto:regs at kinetix.gr> <regs at kinetix.gr
>>     <mailto:regs at kinetix.gr>> wrote:
>>
>>         Hi,
>>
>>            I am making a simple bridge between two call legs :
>>
>>         Client --(a-leg)--> FS --(b-leg)-->Provider
>>
>>         How can I get information like network-address of the Provider,
>>         media-address,
>>         port used, media port used etc. from the second leg (b-leg)?
>>
>>         Is all the information provided by the a-leg available for
>>         the b-leg as
>>         well? If, yese
>>         how can I access it? (and log it to my CDR file eventually)
>>
>>         _______________________________________________
>>         Freeswitch-users mailing list
>>         Freeswitch-users at lists.freeswitch.org
>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Anthony Minessale II
>>
>>     FreeSWITCH http://www.freeswitch.org/
>>     ClueCon http://www.cluecon.com/
>>
>>     AIM: anthm
>>     MSN:anthony_minessale at hotmail.com
>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>>     FreeSWITCH Developer Conference
>>     sip:888 at conference.freeswitch.org
>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>     iax:guest at conference.freeswitch.org/888
>>     <http://iax:guest at conference.freeswitch.org/888>
>>     googletalk:conf+888 at conference.freeswitch.org
>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>     pstn:213-799-1400
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From msc at freeswitch.org  Wed Dec  3 08:04:35 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 3 Dec 2008 08:04:35 -0800
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
	<d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
Message-ID: <3198D996-3558-4539-A1E4-1D3C09835388@freeswitch.org>

Baskar,

Which operating system are you running? I would like to try and  
duplicate symptoms on one of my boxes, all of which run CentOS 5.x

-MC

Sent from my iPhone

On Dec 3, 2008, at 7:32 AM, Baskar <yudha2008 at gmail.com> wrote:

> Hi,
>
> I have newly installed freeswitch in another machine.
>
> After starting the freeswitch I try to dial a extension from console  
> but when i call extension 1002 from freeswitch console, call is  
> connected to extension 1002, but FS is aborted but call is  
> established in1002.
>
> When i dial from console the call get connected and freeswitch is cut.
>
> OUtput:
>
>
> FreeSWITCH Version 1.0.trunk (10567) Started.
> Crash Protection [Disabled]
> Max Sessions[1000]
> Session Rate[30]
> SQL [Enabled]
> 2008-12-03 21:02:21 [CONSOLE] mod_local_stream.c:142  
> read_stream_thread() Can't open directory: /usr/local/freeswitch/ 
> sounds/music/16000
>
> freeswitch at hp30094686650.optimus.co.in> pa devlist
> API CALL [pa(devlist)] output:
> 0;/dev/dsp;16;4
> 1;Intel ICH5: Intel ICH5 (hw:0,0);2;6
> 2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0
> 3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0
> 4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0
> 5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2
> 6;front;0;6
> 7;surround40;0;4
> 8;surround41;0;128
> 9;surround50;0;128
> 10;surround51;0;6
> 11;iec958;0;2
> 12;spdif;0;2
> 13;default;128;128
> 14;dmix;0;2
>
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-03 21:06:11 [NOTICE] switch_channel.c:564  
> switch_channel_set_name() New Channel portaudio/1002  
> [fae97d5b-3480-410e-af0a-192d00710537]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-03 21:06:12 [NOTICE]  
> mod_portaudio.c:1586 place_call() Channel [portaudio/1002] has been  
> answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:fae97d5b-3480-410e-af0a-192d00710537
>
> 2008-12-03 21:06:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()  
> Processing FreeSWITCH->1002 in context default
> 2008-12-03 21:06:12 [WARNING] switch_ivr.c:1840  
> switch_ivr_set_user() can't find user [default@]
> 2008-12-03 21:06:12 [INFO] mod_dptools.c:872 info_function()  
> CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228318571584600]
> Caller-Channel-Created-Time: [1228318571584600]
> Caller-Channel-Answered-Time: [1228318572164620]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-03 21:06:12 [INFO] mod_dptools.c:858 log_function() Answer- 
> State []n
> 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1  
> execute_extension::dx XML features
> 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ 
> usr/local/freeswitch/recordings/0000000000.2008-12-03-21-06-12.wav
> 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3  
> execute_extension::cf XML features
> 2008-12-03 21:06:12 [NOTICE] switch_channel.c:564  
> switch_channel_set_name() New Channel sofia/internal/sip:1002 at 172.20.179.201 
> :37046;rinstance=e6259d34a17a130a [e9a905cd-dc7c-49b1- 
> b3f7-1cd52c1129d1]
> freeswitch: src/switch_core_io.c:202:  
> switch_core_session_read_frame: Assertion `(*frame)->codec != ((void  
> *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
>
> After installing current svn trunk also i get the same error.I cant  
> able to recover the failure .Correct me were i am wrong.
>
>
> Thanks Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dave at 3c.co.uk  Wed Dec  3 08:21:47 2008
From: dave at 3c.co.uk (David Knell)
Date: Wed, 03 Dec 2008 16:21:47 +0000
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <20081203144802.281260@gmx.net>
References: <20081203031552.178560@gmx.net> <49366A6D.4000702@3c.co.uk>
	<20081203144802.281260@gmx.net>
Message-ID: <4936B21B.1030607@3c.co.uk>

Hi Klaus,

There's two differences that I can see between what you're doing and 
what we do:-
1.  We're using the socket in async mode (shouldn't make any difference)
2.  You don't need to send the UUID in after the sendmsg - FS already 
knows which call you're controlling.

Cheers --

Dave
> Hi All,
>
> Thanks for your feedback. I must be doing something fundamentally wrong. Inbound socket is working without problems. But the exact things that i do on inbound socket, i'm not able to replcate them on outbound socket.
>
> The global picture: I have on Xlite registered at extension 1002 and another one at extension 1003.  Then i have an extension 8998 in the default context. Here is the extension definition:
>
>    <extension name="outbound-test">
>         <condition field="destination_number" expression="^8998$" >
>             <action application="socket" data="192.168.50.56:10000 full" />
>         </condition>		
>     </extension>
>
> I use Xlite-1003 to call this extension (8998) and the call is properly notified to the remote Java server. 
>
> Then on the Java side, after receiving the event, i send a CONNECT command: "Connect\n\n"
> The answer from Freeswitch is the state of the channel ( a set of variable, value pair).
>
> Up to this point everything seems normal to me. But then, i try to send an answer command:
>
> sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
> call-command: execute
> execute-app-name: answer
> execute-app-arg:
>
> Freswitch replies with: 
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> But the call is still not answered. Nothing happens on the freeswitch console (Log level DEBUG) and the dialing XLite is still in calling modus.
>
> Then i try bridging the call to 1002:
>
> sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: sofia/internal/1002%192.168.50.94
>    
> Again Freeswitch does answer with:
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> And yet again, nothing is really happening.
>
> What am i missing here?
>
> Thanks,
> Klaus.
>
> -------- Original-Nachricht --------
>   
>> Datum: Wed, 03 Dec 2008 11:15:57 +0000
>> Von: David Knell <dave at 3c.co.uk>
>> An: freeswitch-users at lists.freeswitch.org
>> Betreff: Re: [Freeswitch-users] Bridging from Event Socket API
>>     
>
>   
>> Hi Klaus,
>>
>> Some Perl code snippets - we use:
>> call_command("bridge", "sofia/gateway/bt/$ntd");
>> which, in turn, is:
>> sub call_command($$) {
>>         my $cmd = shift;
>>         my $arg = shift;
>>         print $sock "sendmsg\ncall-command: execute\nexecute-app-name: 
>> $cmd\nexecute-app-arg: $arg\n\n";
>> }
>>
>> Cheers --
>>
>> Dave
>>
>>     
>>> Hi Folks,
>>>
>>> so far i could understand how to bridge calls with Javascript. I'm
>>>       
>> trying to do the same with Java via the Socket Interface. My first trials
>> weren't successful. maybe you can help me understand what is goin on. 
>>     
>>> What i want to do is to bridge an existing leg (Unique-ID is known) to a
>>>       
>> party that wasn't yet dialed (Unique-ID unknown). With javascript it is
>> something like:
>>     
>>> session.bridge("sofia/internal/1002");
>>>
>>> How do i do this using the event socket interface? what
>>>       
>> application/command would i use with which arguments?
>>     
>>> One way i tried to do this is to orginate a call to
>>>       
>> 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it
>> wasn't successful. The only message i had on the FS console is: 
>>     
>>> 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693
>>>       
>> switch_core_session_queue_private_event() Send signal sofia/internal/1001 at 192.168.1.121
>> [BREAK]
>>     
>>> Any idea what i'm missing?
>>>
>>> Thanks,
>>>
>>> Klaus.
>>>
>>>
>>>
>>>
>>>
>>>   
>>>       
>> -- 
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>> http://www.3c.co.uk 
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From klaus.teller at gmx.net  Wed Dec  3 08:43:37 2008
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 03 Dec 2008 17:43:37 +0100
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <4936B21B.1030607@3c.co.uk>
References: <20081203031552.178560@gmx.net> <49366A6D.4000702@3c.co.uk>
	<20081203144802.281260@gmx.net> <4936B21B.1030607@3c.co.uk>
Message-ID: <20081203164337.63380@gmx.net>


> 
> 2.  You don't need to send the UUID in after the sendmsg - FS already 
> knows which call you're controlling.


Bingo! That was it.

Thanks,
Klaus.
-- 
Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL 
f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a


From anthony.minessale at gmail.com  Wed Dec  3 09:13:02 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 11:13:02 -0600
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <4936AA33.7080301@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
	<49369537.6040008@kinetix.gr>
	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
	<4936AA33.7080301@kinetix.gr>
Message-ID: <191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>

It's not an unreasonabe request so i added a patch you can test for me to
trunk that sets network_addr on the reciept of a reply to an invite on an
outbound call. and the 2 variables sip_reply_host and sip_reply_port




local and remote media port reflects the port being used between that leg
and it's remote connection eg the ip and port that the rtp stack was asked
to use.


On Wed, Dec 3, 2008 at 9:48 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:

>  I looked in the b-leg xml cdr and the ip address is not there (for
> signaling) it is only there
> for media (${remote_media_ip}) which is not the same thing now, is it?
>
> While we are at it, I noticed that the ${local_media_port} and
> ${remote_media_port}
> have the same value for each CDR (a or b leg). Shouldn't the first variable
> hold the port
> of the FS (on both legs) and the second variable the port of the client (in
> the a-leg) or the port of
> the provider (in the b-leg)?
>
> Anthony Minessale wrote:
>
> outgoing calls to not have an ip value set.
> if you want to store the dest ip in the cdr you need to set it as a custom
> variable and insert it
> into your template for csv cdr or it will just be there in xml cdr
>
> On Wed, Dec 3, 2008 at 8:18 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:
>
>> b-leg logging is enabled in the cdr module. but in the cdrs I cannot get
>> any variables that refer to the b-leg.
>>
>> I tried the second way using ${sip_to_host} and {bleg_sip_to_host} but :
>>
>> a) the variable returns the FS IP on the a-leg CDR (correctly)
>> b) the variable returns nothing on the b-leg CDR (empty). Shouldn't it
>> return the "to" host of the b-leg (my providers address)?
>>
>>
>> Anthony Minessale wrote:
>>
>>  2 options.
>> 1) enable b-leg logging on the cdr module.
>> 2) you can use the prefix bleg_ in a variable context to get to
>> caller_profile members
>> from the b leg.
>>
>> eg ${bleg_caller_id_name}
>>
>>
>> On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:
>>
>>> Hi,
>>>
>>>    I am making a simple bridge between two call legs :
>>>
>>> Client --(a-leg)--> FS --(b-leg)-->Provider
>>>
>>> How can I get information like network-address of the Provider,
>>> media-address,
>>> port used, media port used etc. from the second leg (b-leg)?
>>>
>>> Is all the information provided by the a-leg available for the b-leg as
>>> well? If, yese
>>> how can I access it? (and log it to my CDR file eventually)
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Dec  3 09:24:28 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 11:24:28 -0600
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
	<d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
Message-ID: <191c3a030812030924u457f934ep77bd70680f583fcd@mail.gmail.com>

please clean all the core.* files

reproduce the problem which will generate a core.xyz file (xyz is some
number)

run the command.

gdb /usr/local/freeswitch/bin/freeswitch core.xzy

when it loads issue the command

bt

and send me the output.








-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gab.tai at xtra.co.nz  Wed Dec  3 09:43:13 2008
From: gab.tai at xtra.co.nz (Gab Tai)
Date: Thu, 4 Dec 2008 06:43:13 +1300
Subject: [Freeswitch-users] Placing call to remote extension
Message-ID: <IJEDKKPJNGCNNCHBNDBHIEMNCBAA.gab.tai@xtra.co.nz>

Hi everyone,

I am Gab and just joined the group.  Also, I am new to FS but want to learn
and delve into the dept as fast as possible.  I have one last mile question
and was hoping I could pick from someone's wealth of knowledge and
understanding of the platform.

I have setup FS with 5 extensions as follows:


  1.. 1 extension [UA(a)] locally registered on the same network  NET(a) as
the realm of the FS(a)
  2.. 2 extensions [UA(b1) & UA(b2) ]  remotely registered to FS(a) from
subnet B
  3.. 2 extensions [UA(c1) & UA(c2) ] remotely registered to FS(a) from
subnet C
  4.. I am not using any provider
Current situation
  a.. All remote extensions can call UA(a) and transfer media (voice)
  b.. UA(a) cannot call remote extensions. Error message "Sofia cannot open
channel,; user not registered". But, please note that the user is actually
registered.
  c.. Remote extensions UA(b1) cannot call UA(b2) and cannot call UA(c1) nor
UA(c2)
  d.. Remote extensions UA(c1) cannot call UA(c2) and cannot call UA(b1) nor
UA(b2)

Need
  a.. How do I place call to remote extensions from local extensions?
  b.. How do I bridge media between 2 remote extensions, registered to FS(a)
from same network or different network?
I hope this is not too much for a starter but would greatly appreciate any
thoughts and/or guidance.

Sincere regards to all.
Gab
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From regs at kinetix.gr  Wed Dec  3 10:20:28 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 20:20:28 +0200
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>
References: <493689D8.9040708@kinetix.gr>	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>	<49369537.6040008@kinetix.gr>	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>	<4936AA33.7080301@kinetix.gr>
	<191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>
Message-ID: <4936CDEC.4010204@kinetix.gr>

I'll try the patch. Thank you for your time.

As for the local and remote media ports :

I have an endpoint with IP xxx.xxx.xxx.xxx and an FS box with IP 
yyy.yyy.yyy.yyy.
In a SIP bridge each side of the call leg between the two boxes will 
pick a udp port in order to send/receive traffic.

In my CDRs (a-leg) when I call the ${remote_media_port} and 
${local_media_port} it returns the same value (e.g. 18841) for both 
endpoints
(yyy.yyy.yyy.yyy and xxx.xxx.xxx.xxx).

In my b-leg CDR (let's say yyy.yyy.yyy.yyy to zzz.zzz.zzz.zzz) both 
variables hold the same value as well but a different
one than the a-leg's (e.g. 19871)

The way I thought it would happen is that each call leg would have a 
pair of different port numbers for the two variables
because :
 yyy would inform xxx that it should use port A
xxx would inform yyy that it should use port B
(that's one pair)
yyy would inform zzz that it should use port C
zzz would inform yyy that it should use port D
(that's another pair)

so for the a-leg : ${local_media_port} = A, ${remote_media_port} = B
for the b=leg : ${local_media_port} = C, ${remote_media_port} = D

Am I missing something?


Anthony Minessale wrote:
> It's not an unreasonabe request so i added a patch you can test for me 
> to trunk that sets network_addr on the reciept of a reply to an invite 
> on an outbound call. and the 2 variables sip_reply_host and sip_reply_port
>
>
>
>
> local and remote media port reflects the port being used between that 
> leg and it's remote connection eg the ip and port that the rtp stack 
> was asked to use.
>
>
> On Wed, Dec 3, 2008 at 9:48 AM, regs at kinetix.gr 
> <mailto:regs at kinetix.gr> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     I looked in the b-leg xml cdr and the ip address is not there (for
>     signaling) it is only there
>     for media (${remote_media_ip}) which is not the same thing now, is it?
>
>     While we are at it, I noticed that the ${local_media_port} and
>     ${remote_media_port}
>     have the same value for each CDR (a or b leg). Shouldn't the first
>     variable hold the port
>     of the FS (on both legs) and the second variable the port of the
>     client (in the a-leg) or the port of
>     the provider (in the b-leg)?
>
>     Anthony Minessale wrote:
>>     outgoing calls to not have an ip value set.
>>     if you want to store the dest ip in the cdr you need to set it as
>>     a custom variable and insert it
>>     into your template for csv cdr or it will just be there in xml cdr
>>
>>     On Wed, Dec 3, 2008 at 8:18 AM, regs at kinetix.gr
>>     <mailto:regs at kinetix.gr> <regs at kinetix.gr
>>     <mailto:regs at kinetix.gr>> wrote:
>>
>>         b-leg logging is enabled in the cdr module. but in the cdrs I
>>         cannot get any variables that refer to the b-leg.
>>
>>         I tried the second way using ${sip_to_host} and
>>         {bleg_sip_to_host} but :
>>
>>         a) the variable returns the FS IP on the a-leg CDR (correctly)
>>         b) the variable returns nothing on the b-leg CDR (empty).
>>         Shouldn't it return the "to" host of the b-leg (my providers
>>         address)?
>>
>>
>>         Anthony Minessale wrote:
>>>         2 options.
>>>         1) enable b-leg logging on the cdr module.
>>>         2) you can use the prefix bleg_ in a variable context to get
>>>         to caller_profile members
>>>         from the b leg.
>>>
>>>         eg ${bleg_caller_id_name}
>>>
>>>
>>>         On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr
>>>         <mailto:regs at kinetix.gr> <regs at kinetix.gr
>>>         <mailto:regs at kinetix.gr>> wrote:
>>>
>>>             Hi,
>>>
>>>                I am making a simple bridge between two call legs :
>>>
>>>             Client --(a-leg)--> FS --(b-leg)-->Provider
>>>
>>>             How can I get information like network-address of the
>>>             Provider,
>>>             media-address,
>>>             port used, media port used etc. from the second leg (b-leg)?
>>>
>>>             Is all the information provided by the a-leg available
>>>             for the b-leg as
>>>             well? If, yese
>>>             how can I access it? (and log it to my CDR file eventually)
>>>
>>>             _______________________________________________
>>>             Freeswitch-users mailing list
>>>             Freeswitch-users at lists.freeswitch.org
>>>             <mailto:Freeswitch-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>         -- 
>>>         Anthony Minessale II
>>>
>>>         FreeSWITCH http://www.freeswitch.org/
>>>         ClueCon http://www.cluecon.com/
>>>
>>>         AIM: anthm
>>>         MSN:anthony_minessale at hotmail.com
>>>         <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>         GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>         <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>         IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>         FreeSWITCH Developer Conference
>>>         sip:888 at conference.freeswitch.org
>>>         <mailto:sip%3A888 at conference.freeswitch.org>
>>>         iax:guest at conference.freeswitch.org/888
>>>         <http://iax:guest at conference.freeswitch.org/888>
>>>         googletalk:conf+888 at conference.freeswitch.org
>>>         <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>         pstn:213-799-1400
>>>         ------------------------------------------------------------------------
>>>         _______________________________________________
>>>         Freeswitch-users mailing list
>>>         Freeswitch-users at lists.freeswitch.org
>>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         _______________________________________________
>>         Freeswitch-users mailing list
>>         Freeswitch-users at lists.freeswitch.org
>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Anthony Minessale II
>>
>>     FreeSWITCH http://www.freeswitch.org/
>>     ClueCon http://www.cluecon.com/
>>
>>     AIM: anthm
>>     MSN:anthony_minessale at hotmail.com
>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>>     FreeSWITCH Developer Conference
>>     sip:888 at conference.freeswitch.org
>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>     iax:guest at conference.freeswitch.org/888
>>     <http://iax:guest at conference.freeswitch.org/888>
>>     googletalk:conf+888 at conference.freeswitch.org
>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>     pstn:213-799-1400
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From lachezar.valchev at gmail.com  Wed Dec  3 10:27:54 2008
From: lachezar.valchev at gmail.com (Lachezar Valchev)
Date: Wed, 3 Dec 2008 20:27:54 +0200
Subject: [Freeswitch-users] CDR generated on maximum sessions reach
Message-ID: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>

Hello everybody,

I am new to the list and I hope I can find some help here, regarding an
issue I am experiencing with the CDRs written by Freeswitch.

The thing is, I am using the "max-sessions" and the "sessions-per-second"
parameters in switch.conf.xml to limit the maximum number of simultaneous
calls, I want to go through my Freeswitch server.

These options are working well, but I was expecting to have CDRs generated
for the calls, that are dropped when the limit is reached.
Unfortunately there is no such one.

My question is: Is there an option, which allows the generation of CDR, when
a call is dropped, because the maximum sessions limit is reached?

If there is no such option, is there any way to achieve it? Probably by
using the the mod_limit module? Can you, please tell me how to do it?

Any kind of advice is welcomed. Thank you in advance.

Regards,
Lachezar
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From msc at freeswitch.org  Wed Dec  3 11:02:29 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Dec 2008 11:02:29 -0800
Subject: [Freeswitch-users] CDR generated on maximum sessions reach
In-Reply-To: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>
References: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>
Message-ID: <87f2f3b90812031102i5e4f8f19q70de5eb7dfbf7959@mail.gmail.com>

On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev <
lachezar.valchev at gmail.com> wrote:

> Hello everybody,
>
> I am new to the list and I hope I can find some help here, regarding an
> issue I am experiencing with the CDRs written by Freeswitch.
>
> The thing is, I am using the "max-sessions" and the "sessions-per-second"
> parameters in switch.conf.xml to limit the maximum number of simultaneous
> calls, I want to go through my Freeswitch server.
>
> These options are working well, but I was expecting to have CDRs generated
> for the calls, that are dropped when the limit is reached.
> Unfortunately there is no such one.
>
> My question is: Is there an option, which allows the generation of CDR,
> when a call is dropped, because the maximum sessions limit is reached?
>
> If there is no such option, is there any way to achieve it? Probably by
> using the the mod_limit module? Can you, please tell me how to do it?
>

Lachezar,

These are good questions! I'll research them and let you know what I find
out.
-MC


>
> Any kind of advice is welcomed. Thank you in advance.
>
> Regards,
> Lachezar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec  3 11:41:04 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Dec 2008 11:41:04 -0800
Subject: [Freeswitch-users] CDR generated on maximum sessions reach
In-Reply-To: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>
References: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>
Message-ID: <87f2f3b90812031141o6292a322j310f8dc0ff41d22e@mail.gmail.com>

On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev <
lachezar.valchev at gmail.com> wrote:

> Hello everybody,
>
> I am new to the list and I hope I can find some help here, regarding an
> issue I am experiencing with the CDRs written by Freeswitch.
>
> The thing is, I am using the "max-sessions" and the "sessions-per-second"
> parameters in switch.conf.xml to limit the maximum number of simultaneous
> calls, I want to go through my Freeswitch server.
>
> These options are working well, but I was expecting to have CDRs generated
> for the calls, that are dropped when the limit is reached.
> Unfortunately there is no such one.
>
> My question is: Is there an option, which allows the generation of CDR,
> when a call is dropped, because the maximum sessions limit is reached?


This is not possible. A CDR cannot be generated without a session, and a
session will not be generated if the max sessions limit has already been
reached...


>
>
> If there is no such option, is there any way to achieve it? Probably by
> using the the mod_limit module? Can you, please tell me how to do it?
>

mod_limit is most definitely your best option at this point. If you haven't
read this yet please do:
http://wiki.freeswitch.org/wiki/Mod_limit

In the example on that page, you have a "limit_exceeded" extension which
would show up in your CDR, or you can set a specific channel variable which
will magically show up in an XML CDR. (You can modify CSV CDRs to have any
custom channel variables as well. See
http://wiki.freeswitch.org/wiki/Cdrwhich refers to the <templates>
section of
conf/autoload_configs/cdr_csv.conf.xml)

Try adding the limit app in your dialplan and have the limit_exceeded
extension as well. You could set the limit really low for the sake of
testing before setting it to the value necessary for your production
deployment.

Let us know how it goes.

-MC


>
> Any kind of advice is welcomed. Thank you in advance.
>
> Regards,
> Lachezar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at suspicious.org  Wed Dec  3 12:01:47 2008
From: krice at suspicious.org (Ken Rice)
Date: Wed, 03 Dec 2008 14:01:47 -0600
Subject: [Freeswitch-users] CDR generated on maximum sessions reach
In-Reply-To: <87f2f3b90812031141o6292a322j310f8dc0ff41d22e@mail.gmail.com>
Message-ID: <C55C41CB.4ACEA%krice@suspicious.org>

From: Michael Collins <msc at freeswitch.org>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Wed, 3 Dec 2008 11:41:04 -0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] CDR generated on maximum sessions reach



On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev
<lachezar.valchev at gmail.com> wrote:
> Hello everybody,
> 
> I am new to the list and I hope I can find some help here, regarding an issue
> I am experiencing with the CDRs written by Freeswitch.
> 
> The thing is, I am using the "max-sessions" and the "sessions-per-second"
> parameters in switch.conf.xml to limit the maximum number of simultaneous
> calls, I want to go through my Freeswitch server.
> 
> These options are working well, but I was expecting to have CDRs generated for
> the calls, that are dropped when the limit is reached.
> Unfortunately there is no such one.
> 
> My question is: Is there an option, which allows the generation of CDR, when a
> call is dropped, because the maximum sessions limit is reached?

This is not possible. A CDR cannot be generated without a session, and a
session will not be generated if the max sessions limit has already been
reached...
 
> 
> 
> If there is no such option, is there any way to achieve it? Probably by using
> the the mod_limit module? Can you, please tell me how to do it?

mod_limit is most definitely your best option at this point. If you haven't
read this yet please do:
http://wiki.freeswitch.org/wiki/Mod_limit

In the example on that page, you have a "limit_exceeded" extension which
would show up in your CDR, or you can set a specific channel variable which
will magically show up in an XML CDR. (You can modify CSV CDRs to have any
custom channel variables as well. See http://wiki.freeswitch.org/wiki/Cdr
which refers to the <templates> section of
conf/autoload_configs/cdr_csv.conf.xml)

Try adding the limit app in your dialplan and have the limit_exceeded
extension as well. You could set the limit really low for the sake of
testing before setting it to the value necessary for your production
deployment.

------SNIP-------

Michael?s advise here is dead on... The whole point of the session limiter
(and SPS limiters) is to keep the box from melting down... Say in the event
of a DoS (or a telemarketer)... If you want to do a soft limit that?s just
want mod_limit is for and will allow you to do 2 things when used in
conjunction with the max_sessions and sps limits... 1) Soft Limit and still
log the calls 2) still have a hard limit that keeps the box for falling over
dead... 

You can do this with a simple step at the top of your dialplan that
mod_limits everything together...

K
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From anthony.minessale at gmail.com  Wed Dec  3 12:11:40 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 14:11:40 -0600
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <4936CDEC.4010204@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
	<49369537.6040008@kinetix.gr>
	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
	<4936AA33.7080301@kinetix.gr>
	<191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>
	<4936CDEC.4010204@kinetix.gr>
Message-ID: <191c3a030812031211q41b69501q5b1b601adf9d0b4d@mail.gmail.com>

looks like a typo in the code.  I guess nobody ever looked at that field
before.
it should be fixed in r10582



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Wed Dec  3 12:33:25 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Dec 2008 12:33:25 -0800
Subject: [Freeswitch-users] Placing call to remote extension
In-Reply-To: <IJEDKKPJNGCNNCHBNDBHIEMNCBAA.gab.tai@xtra.co.nz>
References: <IJEDKKPJNGCNNCHBNDBHIEMNCBAA.gab.tai@xtra.co.nz>
Message-ID: <87f2f3b90812031233l6cb00525geea2a3ede4c045bd@mail.gmail.com>

Hi Gab!

Welcome to FreeSWITCH. Thanks for your questions. I'm trying to learn all of
this stuff and help others so bear with me while I research these and help
you find the answers.

BTW, are you on IRC? you can visit us for realtime help, #freeswitch on
irc.freenode.net
-MC (mercutioviz on irc)

On Wed, Dec 3, 2008 at 9:43 AM, Gab Tai <gab.tai at xtra.co.nz> wrote:

>  Hi everyone,
>
> I am Gab and just joined the group.  Also, I am new to FS but want to learn
> and delve into the dept as fast as possible.  I have one last mile question
> and was hoping I could pick from someone's wealth of knowledge and
> understanding of the platform.
>
> I have setup FS with 5 extensions as follows:
>
>    1. 1 extension [UA(a)] locally registered on the same network  NET(a)
>    as the realm of the FS(a)
>    2. 2 extensions [UA(b1) & UA(b2) ]  remotely registered to FS(a) from
>    subnet B
>    3. 2 extensions [UA(c1) & UA(c2) ] remotely registered to FS(a) from
>    subnet C
>    4. I am not using any provider
>
> *Current situation*
>
>    - All remote extensions can call UA(a) and transfer media (voice)
>    - UA(a) cannot call remote extensions. Error message "Sofia cannot open
>    channel,; user not registered". But, please note that the user is actually
>    registered.
>    - Remote extensions UA(b1) cannot call UA(b2) and cannot call UA(c1)
>    nor UA(c2)
>    - Remote extensions UA(c1) cannot call UA(c2) and cannot call UA(b1)
>    nor UA(b2)
>
>
> *Need*
>
>    - How do I place call to remote extensions from local extensions?
>    - How do I bridge media between 2 remote extensions, registered to
>    FS(a) from same network or different network?
>
> I hope this is not too much for a starter but would greatly appreciate any
> thoughts and/or guidance.
>
> Sincere regards to all.
> Gab
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jkr888 at gmail.com  Wed Dec  3 12:48:04 2008
From: jkr888 at gmail.com (Johny Kadarisman)
Date: Wed, 3 Dec 2008 15:48:04 -0500
Subject: [Freeswitch-users] Redirect calls between FS
In-Reply-To: <b081cc8f0811290318h5b27af05h531cd80230dd04d2@mail.gmail.com>
References: <b081cc8f0811290318h5b27af05h531cd80230dd04d2@mail.gmail.com>
Message-ID: <b0a271780812031248x4fd7c240rd97ad16bd6ecb34b@mail.gmail.com>

Is these apps will work for you?

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect


On Sat, Nov 29, 2008 at 6:18 AM, Barray McKee <barraymckee at gmail.com> wrote:
> Hello,
>
> I am implementing 2 load balancing FS behind a pair of sip proxies.  Since
> traffic is routed to one of the two FS on a round-robin basis, I need a
> mechanism to reroute call from one FS to another FS under a specific special
> circumstance.  I am think to use the bridge command with bypass_media =
> true, so that the call can just pass through one FS and reach the second
> FS.   Is this the approach that I should use?  I am wondering if there is a
> better approach out there that I just can't think of.
>
> Thanks in advance,
> Barray
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From regs at kinetix.gr  Wed Dec  3 13:24:18 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 23:24:18 +0200
Subject: [Freeswitch-users] How to get info from the b-leg [PATCHED -
 FIXED]
In-Reply-To: <191c3a030812031211q41b69501q5b1b601adf9d0b4d@mail.gmail.com>
References: <493689D8.9040708@kinetix.gr>	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>	<49369537.6040008@kinetix.gr>	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>	<4936AA33.7080301@kinetix.gr>	<191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>	<4936CDEC.4010204@kinetix.gr>
	<191c3a030812031211q41b69501q5b1b601adf9d0b4d@mail.gmail.com>
Message-ID: <4936F902.1070905@kinetix.gr>

I tested both patches from the trunk : network_addr is set to the remote 
IP on the b-leg
and local media port and remote media port hold the correct values when 
called.
Both pathces work like a charm!
Thanks for your time and effort :)

Anthony Minessale wrote:
>
>
> looks like a typo in the code.  I guess nobody ever looked at that 
> field before.
> it should be fixed in r10582
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From msc at freeswitch.org  Wed Dec  3 13:32:25 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Dec 2008 13:32:25 -0800
Subject: [Freeswitch-users] How to get info from the b-leg [PATCHED -
	FIXED]
In-Reply-To: <4936F902.1070905@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
	<49369537.6040008@kinetix.gr>
	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
	<4936AA33.7080301@kinetix.gr>
	<191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>
	<4936CDEC.4010204@kinetix.gr>
	<191c3a030812031211q41b69501q5b1b601adf9d0b4d@mail.gmail.com>
	<4936F902.1070905@kinetix.gr>
Message-ID: <87f2f3b90812031332t291a96d5heafbfc2914e48f09@mail.gmail.com>

And thank you for testing and being gracious! :)
-MC

On Wed, Dec 3, 2008 at 1:24 PM, regs at kinetix.gr <regs at kinetix.gr> wrote:

>  I tested both patches from the trunk : network_addr is set to the remote
> IP on the b-leg
> and local media port and remote media port hold the correct values when
> called.
> Both pathces work like a charm!
> Thanks for your time and effort :)
>
> Anthony Minessale wrote:
>
>
>
> looks like a typo in the code.  I guess nobody ever looked at that field
> before.
> it should be fixed in r10582
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kkielhofner at star2star.com  Wed Dec  3 14:26:15 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Wed, 3 Dec 2008 17:26:15 -0500
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <87f2f3b90812021718j5aae69aav6dd8ee7953e2b1ff@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
	<87f2f3b90812021718j5aae69aav6dd8ee7953e2b1ff@mail.gmail.com>
Message-ID: <2d9149cd0812031426h71ccaddqfbd6d6af34e8eaa7@mail.gmail.com>

On Tue, Dec 2, 2008 at 8:18 PM, Michael Collins <msc at freeswitch.org> wrote:
> Kristian,
>
> Are you on the IRC channel by any chance?
> -MC (IRC: mercutioviz)
>

Me? Never!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From sobolewski at gmail.com  Wed Dec  3 14:33:24 2008
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Wed, 3 Dec 2008 23:33:24 +0100
Subject: [Freeswitch-users] screen_bit is always true
Message-ID: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>

Hi

I was trying  to create extension in which I would check whether
privacy=full in Remote-Party-ID header is set.
So I made this.

   <condition field="${screen_bit}" expression="^true$" break="never">
          <action application="set"
data="origination_caller_id_number=Anonymous"/>
   </condition>

But screen_bit is always true, regardless RPID privacy value.

        mod_dialplan_xml.c:117 parse_exten() Regex: [tooser]
${screen_bit}(true) =~ /^true$/

As it wasn't strange enough, in cdr_csv screen_bit take "true" or
"false" (as it should).

Is it a bug or me doing something wrong?

--
regards
Piotr Sobolewski


From krice at suspicious.org  Wed Dec  3 16:14:57 2008
From: krice at suspicious.org (Ken Rice)
Date: Wed, 03 Dec 2008 18:14:57 -0600
Subject: [Freeswitch-users] screen_bit is always true
In-Reply-To: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
Message-ID: <C55C7D21.4AD00%krice@suspicious.org>

The screen bit is a trust bit... ie: do we trust the RPID we got from the
upstream or not

K


> From: Piotr Sobolewski <sobolewski at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 3 Dec 2008 23:33:24 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] screen_bit is always true
> 
> Hi
> 
> I was trying  to create extension in which I would check whether
> privacy=full in Remote-Party-ID header is set.
> So I made this.
> 
>    <condition field="${screen_bit}" expression="^true$" break="never">
>           <action application="set"
> data="origination_caller_id_number=Anonymous"/>
>    </condition>
> 
> But screen_bit is always true, regardless RPID privacy value.
> 
>         mod_dialplan_xml.c:117 parse_exten() Regex: [tooser]
> ${screen_bit}(true) =~ /^true$/
> 
> As it wasn't strange enough, in cdr_csv screen_bit take "true" or
> "false" (as it should).
> 
> Is it a bug or me doing something wrong?
> 
> --
> regards
> Piotr Sobolewski
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Wed Dec  3 17:38:13 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Dec 2008 19:38:13 -0600
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <2d9149cd0812031426h71ccaddqfbd6d6af34e8eaa7@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
	<87f2f3b90812021718j5aae69aav6dd8ee7953e2b1ff@mail.gmail.com>
	<2d9149cd0812031426h71ccaddqfbd6d6af34e8eaa7@mail.gmail.com>
Message-ID: <BC166D25-7051-4E3F-B27B-DE55984A1E63@freeswitch.org>

You scared? muhahaha  Will I see you at ClueCon 09, its the first week  
in Aug. again.... in Chicago.

/b

On Dec 3, 2008, at 4:26 PM, Kristian Kielhofner wrote:

> Me? Never!



From brian at freeswitch.org  Wed Dec  3 17:38:53 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Dec 2008 19:38:53 -0600
Subject: [Freeswitch-users] screen_bit is always true
In-Reply-To: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
References: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
Message-ID: <A914D032-DC2F-4DF8-9808-4DE6EBF9A08E@freeswitch.org>

I think we covered this on IRC already didn't we?

/b

On Dec 3, 2008, at 4:33 PM, Piotr Sobolewski wrote:

>
> Is it a bug or me doing something wrong?



From sobolewski at gmail.com  Wed Dec  3 18:02:40 2008
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Thu, 4 Dec 2008 03:02:40 +0100
Subject: [Freeswitch-users] screen_bit is always true
In-Reply-To: <C55C7D21.4AD00%krice@suspicious.org>
References: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
	<C55C7D21.4AD00%krice@suspicious.org>
Message-ID: <3666ca0d0812031802o754ed22eub5435345e542153e@mail.gmail.com>

On Thu, Dec 4, 2008 at 1:14 AM, Ken Rice <krice at suspicious.org> wrote:
> The screen bit is a trust bit... ie: do we trust the RPID we got from the
> upstream or not
>
> K

I had privacy_hide_number in cdr_csv and I was thinking it was
screen_bit, all that confused me.
Now I understand where I was wrong.

BTW: is there a way to remove RPID header?


-- 
Piotr Sobolewski
sobolewski at gmail.com


From brian at freeswitch.org  Wed Dec  3 18:12:27 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Dec 2008 20:12:27 -0600
Subject: [Freeswitch-users] screen_bit is always true
In-Reply-To: <3666ca0d0812031802o754ed22eub5435345e542153e@mail.gmail.com>
References: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
	<C55C7D21.4AD00%krice@suspicious.org>
	<3666ca0d0812031802o754ed22eub5435345e542153e@mail.gmail.com>
Message-ID: <4F994CD4-43CF-4AC9-8EC2-A66AD715A1B0@freeswitch.org>

I already told you this one on IRC too :P  email is too slow today :)

/b

On Dec 3, 2008, at 8:02 PM, Piotr Sobolewski wrote:

> BTW: is there a way to remove RPID header?



From ack at telefonica.net  Wed Dec  3 17:03:08 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Thu, 04 Dec 2008 02:03:08 +0100
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
Message-ID: <1228352588.25709.42.camel@develop4>

Hi guys, 
 
 I've a strange issue with FS , version svn -r10584 ,
when FS bridges a call first 3 seconds of audio are missing , looks that
only happens on PSTN calls and using ringback or transfer_ringback. This
only happens in calls from PSTN , not from VOIP. Some scenarios i tried
to isolate this issue :


- Issue

PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN

- Good setting bypass_media before run bridge but i need rtp in FS path

PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN

- Good 

PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback -> PSTN

- Always good

VOIP --> FS ( brigde ) -> PSTN


Dialplan has nothing wrong ( i guess ):

<extension name="Transfers">
    <condition field="destination_number" expression="^1??XXXXXXXXXX$">
      <action application="answer"/>
      <action application="speak" data="cepstral|Allison-8kHz|blah"/>
      <action application="set" data="hangup_after_bridge=false"/>
      <action application="set" data="playback_terminators=#"/>
      <action application="set" data="ringback=$${us-ring}"/>
      <action application="set" data="transfer_ringback=
$${hold_music}"/>
      <action application="set" data="effective_caller_id_name=
${caller_id_name}"/>
      <action application="set" data="effective_caller_id_number=
${caller_id_number}"/>
      <action application="set" data="originate_timeout=30"/>
      <action application="set" data="call_timeout=30"/>
      <action application="bridge"
data="sofia/default/18008226235 at PSTN_GW"/>
      <action application="speak" data="cepstral|Allison-8kHz|Transfer
finished"/>
      <action application="hangup"/>
    </condition>
  </extension>



Any ideas ?

Attached log of FS ( F8 from console ).


Thanks in advance !

-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

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From anthony.minessale at gmail.com  Wed Dec  3 20:12:00 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 22:12:00 -0600
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <1228352588.25709.42.camel@develop4>
References: <1228352588.25709.42.camel@develop4>
Message-ID: <191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>

what does PSTN represent?

I know what the PSTN is but how are you reaching it?
is it TDM, SIP etc... what gateway type other details.


On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <ack at telefonica.net> wrote:

> Hi guys,
>
>  I've a strange issue with FS , version svn -r10584 ,
> when FS bridges a call first 3 seconds of audio are missing , looks that
> only happens on PSTN calls and using ringback or transfer_ringback. This
> only happens in calls from PSTN , not from VOIP. Some scenarios i tried
> to isolate this issue :
>
>
> - Issue
>
> PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>
> - Good setting bypass_media before run bridge but i need rtp in FS path
>
> PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>
> - Good
>
> PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback -> PSTN
>
> - Always good
>
> VOIP --> FS ( brigde ) -> PSTN
>
>
> Dialplan has nothing wrong ( i guess ):
>
> <extension name="Transfers">
>    <condition field="destination_number" expression="^1??XXXXXXXXXX$">
>      <action application="answer"/>
>      <action application="speak" data="cepstral|Allison-8kHz|blah"/>
>      <action application="set" data="hangup_after_bridge=false"/>
>      <action application="set" data="playback_terminators=#"/>
>      <action application="set" data="ringback=$${us-ring}"/>
>      <action application="set" data="transfer_ringback=
> $${hold_music}"/>
>      <action application="set" data="effective_caller_id_name=
> ${caller_id_name}"/>
>      <action application="set" data="effective_caller_id_number=
> ${caller_id_number}"/>
>      <action application="set" data="originate_timeout=30"/>
>      <action application="set" data="call_timeout=30"/>
>      <action application="bridge"
> data="sofia/default/18008226235 at PSTN_GW"/>
>      <action application="speak" data="cepstral|Allison-8kHz|Transfer
> finished"/>
>      <action application="hangup"/>
>    </condition>
>  </extension>
>
>
>
> Any ideas ?
>
> Attached log of FS ( F8 from console ).
>
>
> Thanks in advance !
>
> --
> Angel Carpintero
> ack ( at ) telefonica ( dot ) net
>
> Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ack at telefonica.net  Wed Dec  3 20:46:21 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Thu, 04 Dec 2008 05:46:21 +0100
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
References: <1228352588.25709.42.camel@develop4>
	<191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
Message-ID: <1228365981.25709.60.camel@develop4>

No TDM , all is SIP :


PSTN ---> Sip Proxy_A --> FS ( brigde ) ringback/transfer_ringback 
-> Sip Proxy_B --> PSTN 


In logfile i think you can get some details about Media Gateways
( Sonus ) PSTN inbound / outbound is provided by Level3. 

I can get a capture of a call if you want, in capture the audio is not 
missing, issue with :

- rtp buffer ?
- Sonus ? 

Let me know anything you need so i can provide a log or create a new
scenario.


Thanks,

El mi?, 03-12-2008 a las 22:12 -0600, Anthony Minessale escribi?:
> what does PSTN represent? 
> 
> I know what the PSTN is but how are you reaching it?
> is it TDM, SIP etc... what gateway type other details.
> 
> 
> On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <ack at telefonica.net>
> wrote:
>         Hi guys,
>         
>          I've a strange issue with FS , version svn -r10584 ,
>         when FS bridges a call first 3 seconds of audio are missing ,
>         looks that
>         only happens on PSTN calls and using ringback or
>         transfer_ringback. This
>         only happens in calls from PSTN , not from VOIP. Some
>         scenarios i tried
>         to isolate this issue :
>         
>         
>         - Issue
>         
>         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>         
>         - Good setting bypass_media before run bridge but i need rtp
>         in FS path
>         
>         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>         
>         - Good
>         
>         PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback ->
>         PSTN
>         
>         - Always good
>         
>         VOIP --> FS ( brigde ) -> PSTN
>         
>         
>         Dialplan has nothing wrong ( i guess ):
>         
>         <extension name="Transfers">
>            <condition field="destination_number"
>         expression="^1??XXXXXXXXXX$">
>              <action application="answer"/>
>              <action application="speak" data="cepstral|Allison-8kHz|
>         blah"/>
>              <action application="set"
>         data="hangup_after_bridge=false"/>
>              <action application="set" data="playback_terminators=#"/>
>              <action application="set" data="ringback=$${us-ring}"/>
>              <action application="set" data="transfer_ringback=
>         $${hold_music}"/>
>              <action application="set" data="effective_caller_id_name=
>         ${caller_id_name}"/>
>              <action application="set"
>         data="effective_caller_id_number=
>         ${caller_id_number}"/>
>              <action application="set" data="originate_timeout=30"/>
>              <action application="set" data="call_timeout=30"/>
>              <action application="bridge"
>         data="sofia/default/18008226235 at PSTN_GW"/>
>              <action application="speak" data="cepstral|Allison-8kHz|
>         Transfer
>         finished"/>
>              <action application="hangup"/>
>            </condition>
>          </extension>
>         
>         
>         
>         Any ideas ?
>         
>         Attached log of FS ( F8 from console ).
>         
>         
>         Thanks in advance !
>         
>         --
>         Angel Carpintero
>         ack ( at ) telefonica ( dot ) net
>         
>         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
>         6EF1 B90D
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400

-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

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From pieter_eduard at biznetnetworks.com  Thu Dec  4 00:03:36 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Thu, 04 Dec 2008 15:03:36 +0700
Subject: [Freeswitch-users] freeswitch enum
Message-ID: <49378ED8.9010404@biznetnetworks.com>

Hi,

I'm trying to query freeswitch  to use my bind base enum server but i'm 
having trouble to query the enum from Freeswitch CLI.

this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after 
restart the dns, i have this :

[root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
Using domain server:
Name: localhost
Address: 127.0.0.1#53
Aliases:

8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel" 
"!^.*$!tel:\\12345678!" .
8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip" 
"!^.*$!sip:1000 at test.com!" .

and if use the ip, the result is the same :
[root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
Using domain server:
Name: 1.2.3.4
Address: 1.2.3.4#53
Aliases:

8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip" 
"!^.*$!sip:1000 at test.com!" .
8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel" 
"!^.*$!tel:\\12345678!" .

But if i try to query from the freeswitch CLI that's installed in the 
same box, i get this :

freeswitch at fsbox> enum 12345678 localhost
API CALL [enum(12345678 localhost)] output:
No Match!

freeswitch at fsbox> enum 12345678 1.2.3.4
API CALL [enum(12345678 1.2.3.4)] output:
No Match!

even if i change the enum.conf.xml dns root with my enum ip and reload 
the freeswitch i still get the same error.

Can anyone help me on this? i just want the fs to query on my local enum 
or to query on different enum server.

Regards,
-Pieter-



From krice at suspicious.org  Thu Dec  4 00:13:08 2008
From: krice at suspicious.org (Ken Rice)
Date: Thu, 04 Dec 2008 02:13:08 -0600
Subject: [Freeswitch-users] freeswitch enum
In-Reply-To: <49378ED8.9010404@biznetnetworks.com>
Message-ID: <C55CED34.4AD19%krice@suspicious.org>

Ok 1) Overriding the e164.arpa is probably not a good Idea... 2) the second
param for the enum command is a domain not a Server IP address

Example: enum 1234567890 e164.org

You should probably set up your enum records on your own private domain or
use a real domain that you own... If you have to use something like e164.int
as the domain then tell your name server its the SOA for that domain and set
all your records up there...

Then in FreeSwitch tell it the default domain is e164.int

Chances are if you do something like enum 18005551212 e164.org it will work
correctly...


> From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 04 Dec 2008 15:03:36 +0700
> To: "freeswitch-users at lists.freeswitch.org"
> <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] freeswitch enum
> 
> Hi,
> 
> I'm trying to query freeswitch  to use my bind base enum server but i'm
> having trouble to query the enum from Freeswitch CLI.
> 
> this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after
> restart the dns, i have this :
> 
> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
> Using domain server:
> Name: localhost
> Address: 127.0.0.1#53
> Aliases:
> 
> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
> "!^.*$!tel:\\12345678!" .
> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
> "!^.*$!sip:1000 at test.com!" .
> 
> and if use the ip, the result is the same :
> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
> Using domain server:
> Name: 1.2.3.4
> Address: 1.2.3.4#53
> Aliases:
> 
> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
> "!^.*$!sip:1000 at test.com!" .
> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
> "!^.*$!tel:\\12345678!" .
> 
> But if i try to query from the freeswitch CLI that's installed in the
> same box, i get this :
> 
> freeswitch at fsbox> enum 12345678 localhost
> API CALL [enum(12345678 localhost)] output:
> No Match!
> 
> freeswitch at fsbox> enum 12345678 1.2.3.4
> API CALL [enum(12345678 1.2.3.4)] output:
> No Match!
> 
> even if i change the enum.conf.xml dns root with my enum ip and reload
> the freeswitch i still get the same error.
> 
> Can anyone help me on this? i just want the fs to query on my local enum
> or to query on different enum server.
> 
> Regards,
> -Pieter-
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From carole.olivier at enst.fr  Thu Dec  4 00:28:04 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 4 Dec 2008 00:28:04 -0800 (PST)
Subject: [Freeswitch-users] re gistration and calls through different user
 agent
Message-ID: <20829148.post@talk.nabble.com>


Hello,

I have recently installed freeswitch on Opensuse 10.3.

I have a question about the sofia agents which are already present in the
default installation:
I have 2 snom phones (each with a user1001 and user1002), I configured one
such that it registered on the default port 5060 (so on the user agent
internal) and the other one on the port 5080 (on the external user agent).
The aim was to see what happen.

Since both were able to register I guess that the directory
/directory/default/ is used by both sofia agent else user1002 should not be
able to register, isn't? I am confused here because I would have said that
the directory /directory/default/ was just for the users that register on
internal...

I had a second problem: if user1001 tries to call user1002 it did not work
(but the opposite is ok). Has this something to do with the dialplan? 

Thanks,
Carole
-- 
View this message in context: http://www.nabble.com/registration-and-calls-through-different-user-agent-tp20829148p20829148.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From pieter_eduard at biznetnetworks.com  Thu Dec  4 01:12:36 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Thu, 04 Dec 2008 16:12:36 +0700
Subject: [Freeswitch-users] freeswitch enum
In-Reply-To: <C55CED34.4AD19%krice@suspicious.org>
References: <C55CED34.4AD19%krice@suspicious.org>
Message-ID: <49379F04.10704@biznetnetworks.com>

Ken,

I have done what you suggested bellow, for security reason i can not 
paste the query using the domain, just the localhost and i change the 
public  ip to 1.2.3.4.
So i already setup a real name server with  real domain and it uses  
public ip, configure the 7.6.5.4.3.2.1.e164.arpa as a domain with my 
name server as  SOA and NS record and yet the freeswitch CLI still give 
me no match when i try to query it using my name server.

And yes, if i query 18005551212 e164.org it works like a charm ;-)

Any other suggestions?

Regards,

-Pieter-
I
Ken Rice wrote:
> Ok 1) Overriding the e164.arpa is probably not a good Idea... 2) the second
> param for the enum command is a domain not a Server IP address
>
> Example: enum 1234567890 e164.org
>
> You should probably set up your enum records on your own private domain or
> use a real domain that you own... If you have to use something like e164.int
> as the domain then tell your name server its the SOA for that domain and set
> all your records up there...
>
> Then in FreeSwitch tell it the default domain is e164.int
>
> Chances are if you do something like enum 18005551212 e164.org it will work
> correctly...
>
>
>   
>> From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Thu, 04 Dec 2008 15:03:36 +0700
>> To: "freeswitch-users at lists.freeswitch.org"
>> <freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] freeswitch enum
>>
>> Hi,
>>
>> I'm trying to query freeswitch  to use my bind base enum server but i'm
>> having trouble to query the enum from Freeswitch CLI.
>>
>> this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after
>> restart the dns, i have this :
>>
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
>> Using domain server:
>> Name: localhost
>> Address: 127.0.0.1#53
>> Aliases:
>>
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>>
>> and if use the ip, the result is the same :
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
>> Using domain server:
>> Name: 1.2.3.4
>> Address: 1.2.3.4#53
>> Aliases:
>>
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>>
>> But if i try to query from the freeswitch CLI that's installed in the
>> same box, i get this :
>>
>> freeswitch at fsbox> enum 12345678 localhost
>> API CALL [enum(12345678 localhost)] output:
>> No Match!
>>
>> freeswitch at fsbox> enum 12345678 1.2.3.4
>> API CALL [enum(12345678 1.2.3.4)] output:
>> No Match!
>>
>> even if i change the enum.conf.xml dns root with my enum ip and reload
>> the freeswitch i still get the same error.
>>
>> Can anyone help me on this? i just want the fs to query on my local enum
>> or to query on different enum server.
>>
>> Regards,
>> -Pieter-
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> .
>
>   
-------------- next part --------------
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From krice at suspicious.org  Thu Dec  4 01:26:33 2008
From: krice at suspicious.org (Ken Rice)
Date: Thu, 04 Dec 2008 03:26:33 -0600
Subject: [Freeswitch-users] freeswitch enum
In-Reply-To: <49379F04.10704@biznetnetworks.com>
Message-ID: <C55CFE69.4AD1F%krice@suspicious.org>

No other suggestion your DNS setup is broken... You have proven that
freeswitch is working by querying e164.org...

Again you need to use a domain other than one that ends in .arpa that is a
reservered TLD and you will break things... Use the .int TLD that?s what its
for and if you are really worried about leaking things to the outside world
put an ACL on your DNS server to not let anyone outside your netblocks query
it... 2) resolv.conf on your fs box must point only at your name servers
that are SOA (either primary or secondary) for your private domain and much
not contain servers of your ISP that will cause lookup failures...
3) and once again you do not specify the IP address you are querying using
the enum command... You specify the number and the root (root being the
domain ie: mydomain.int)

Beyond that you need to check out other resources appropriate for setting up
enum records for your specific DNS implementation



From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 04 Dec 2008 16:12:36 +0700
To: "freeswitch-users at lists.freeswitch.org"
<freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch enum

Ken,

I have done what you suggested bellow, for security reason i can not paste
the query using the domain, just the localhost and i change the public  ip
to 1.2.3.4.
So i already setup a real name server with  real domain and it uses public
ip, configure the 7.6.5.4.3.2.1.e164.arpa as a domain with my name server as
SOA and NS record and yet the freeswitch CLI still give me no match when i
try to query it using my name server.

And yes, if i query 18005551212 e164.org it works like a charm ;-)

Any other suggestions?

Regards,

-Pieter-
I
Ken Rice wrote: 
>  
> Ok 1) Overriding the e164.arpa is probably not a good Idea... 2) the second
> param for the enum command is a domain not a Server IP address
> 
> Example: enum 1234567890 e164.org
> 
> You should probably set up your enum records on your own private domain or
> use a real domain that you own... If you have to use something like e164.int
> as the domain then tell your name server its the SOA for that domain and set
> all your records up there...
> 
> Then in FreeSwitch tell it the default domain is e164.int
> 
> Chances are if you do something like enum 18005551212 e164.org it will work
> correctly...
> 
> 
>   
>  
>>  
>> From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
>> <mailto:pieter_eduard at biznetnetworks.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> Date: Thu, 04 Dec 2008 15:03:36 +0700
>> To: "freeswitch-users at lists.freeswitch.org"
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] freeswitch enum
>> 
>> Hi,
>> 
>> I'm trying to query freeswitch  to use my bind base enum server but i'm
>> having trouble to query the enum from Freeswitch CLI.
>> 
>> this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after
>> restart the dns, i have this :
>> 
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
>> Using domain server:
>> Name: localhost
>> Address: 127.0.0.1#53
>> Aliases:
>> 
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 
>> and if use the ip, the result is the same :
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
>> Using domain server:
>> Name: 1.2.3.4
>> Address: 1.2.3.4#53
>> Aliases:
>> 
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 
>> But if i try to query from the freeswitch CLI that's installed in the
>> same box, i get this :
>> 
>> freeswitch at fsbox> enum 12345678 localhost
>> API CALL [enum(12345678 localhost)] output:
>> No Match!
>> 
>> freeswitch at fsbox> enum 12345678 1.2.3.4
>> API CALL [enum(12345678 1.2.3.4)] output:
>> No Match!
>> 
>> even if i change the enum.conf.xml dns root with my enum ip and reload
>> the freeswitch i still get the same error.
>> 
>> Can anyone help me on this? i just want the fs to query on my local enum
>> or to query on different enum server.
>> 
>> Regards,
>> -Pieter-
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>>  
>>  
>> 
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> .
>> 
>>   
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

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From krice at suspicious.org  Thu Dec  4 01:51:05 2008
From: krice at suspicious.org (Ken Rice)
Date: Thu, 04 Dec 2008 03:51:05 -0600
Subject: [Freeswitch-users] freeswitch enum
In-Reply-To: <C55CFE69.4AD1F%krice@suspicious.org>
Message-ID: <C55D0429.4AD21%krice@suspicious.org>

Ooops lemme correct myself...  .int is for international orgs and you
shouldn?t use that either... Try .localnet or .rofl something that is not
and probably never will be allocated.....

K



From: Ken Rice <krice at suspicious.org>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 04 Dec 2008 03:26:33 -0600
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch enum

No other suggestion your DNS setup is broken... You have proven that
freeswitch is working by querying e164.org...

Again you need to use a domain other than one that ends in .arpa that is a
reservered TLD and you will break things... Use the .int TLD that?s what its
for and if you are really worried about leaking things to the outside world
put an ACL on your DNS server to not let anyone outside your netblocks query
it... 2) resolv.conf on your fs box must point only at your name servers
that are SOA (either primary or secondary) for your private domain and much
not contain servers of your ISP that will cause lookup failures...
3) and once again you do not specify the IP address you are querying using
the enum command... You specify the number and the root (root being the
domain ie: mydomain.int)

Beyond that you need to check out other resources appropriate for setting up
enum records for your specific DNS implementation



From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 04 Dec 2008 16:12:36 +0700
To: "freeswitch-users at lists.freeswitch.org"
<freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch enum

Ken,

I have done what you suggested bellow, for security reason i can not paste
the query using the domain, just the localhost and i change the public  ip
to 1.2.3.4.
So i already setup a real name server with  real domain and it uses public
ip, configure the 7.6.5.4.3.2.1.e164.arpa as a domain with my name server as
SOA and NS record and yet the freeswitch CLI still give me no match when i
try to query it using my name server.

And yes, if i query 18005551212 e164.org it works like a charm ;-)

Any other suggestions?

Regards,

-Pieter-
I
Ken Rice wrote: 
>  
> Ok 1) Overriding the e164.arpa is probably not a good Idea... 2) the second
> param for the enum command is a domain not a Server IP address
> 
> Example: enum 1234567890 e164.org
> 
> You should probably set up your enum records on your own private domain or
> use a real domain that you own... If you have to use something like e164.int
> as the domain then tell your name server its the SOA for that domain and set
> all your records up there...
> 
> Then in FreeSwitch tell it the default domain is e164.int
> 
> Chances are if you do something like enum 18005551212 e164.org it will work
> correctly...
> 
> 
>   
>  
>>  
>> From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
>> <mailto:pieter_eduard at biznetnetworks.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> Date: Thu, 04 Dec 2008 15:03:36 +0700
>> To: "freeswitch-users at lists.freeswitch.org"
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] freeswitch enum
>> 
>> Hi,
>> 
>> I'm trying to query freeswitch  to use my bind base enum server but i'm
>> having trouble to query the enum from Freeswitch CLI.
>> 
>> this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after
>> restart the dns, i have this :
>> 
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
>> Using domain server:
>> Name: localhost
>> Address: 127.0.0.1#53
>> Aliases:
>> 
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 
>> and if use the ip, the result is the same :
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
>> Using domain server:
>> Name: 1.2.3.4
>> Address: 1.2.3.4#53
>> Aliases:
>> 
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 
>> But if i try to query from the freeswitch CLI that's installed in the
>> same box, i get this :
>> 
>> freeswitch at fsbox> enum 12345678 localhost
>> API CALL [enum(12345678 localhost)] output:
>> No Match!
>> 
>> freeswitch at fsbox> enum 12345678 1.2.3.4
>> API CALL [enum(12345678 1.2.3.4)] output:
>> No Match!
>> 
>> even if i change the enum.conf.xml dns root with my enum ip and reload
>> the freeswitch i still get the same error.
>> 
>> Can anyone help me on this? i just want the fs to query on my local enum
>> or to query on different enum server.
>> 
>> Regards,
>> -Pieter-
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>>  
>>  
>> 
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> .
>> 
>>   
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

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From cstomi.levlist at gmail.com  Thu Dec  4 03:22:04 2008
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Thu, 04 Dec 2008 12:22:04 +0100
Subject: [Freeswitch-users] voicemail disk quota poll
Message-ID: <4937BD5C.7050002@gmail.com>

Hello,

I would like to use a disk quota in users' voicemail
(http://jira.freeswitch.org/browse/MODAPP-173)

We haven't decided yet what would be the better prompts to play to the 
caller when the mailbox is full.
Please advice some messages!

Thanks in advance,
Tamas




From faisalmaqsoodi at yahoo.com  Thu Dec  4 03:37:09 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Thu, 4 Dec 2008 03:37:09 -0800 (PST)
Subject: [Freeswitch-users] mod_spidermonkey()  Session is not active!
Message-ID: <966608.77509.qm@web30706.mail.mud.yahoo.com>


Hi
I m trying to use a Javascript application which uses? mod_pocketsphinx and mod_spidermonkey. Both r loaded but this error msg is displayed:
2008-12-04 16:31:09 [ERR] inline:1 mod_spidermonkey()? Session is not active!
How can i remove this error?
??????????????????????????????????? Faisal



      
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From saigop at gmail.com  Thu Dec  4 04:50:51 2008
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Thu, 4 Dec 2008 18:20:51 +0530
Subject: [Freeswitch-users] Predictive Dialing
Message-ID: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>

Hi,
   I would like to have predictive dialing. In asterisk we used manager api
and for outbound we use originate. The originate command will dial a number
where asterisk answer the call and then we predict the answering machine
with the silence file. Inspite of that human voice is detected and
transferred to agents.

   Like this how can we go ahead with Freeswitch? Any help would be
appreciated.

-- 
Thank you  with regards,
Gopal,
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From brian at freeswitch.org  Thu Dec  4 06:23:45 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2008 08:23:45 -0600
Subject: [Freeswitch-users] re gistration and calls through different
	user agent
In-Reply-To: <20829148.post@talk.nabble.com>
References: <20829148.post@talk.nabble.com>
Message-ID: <6643DA4A-019C-4446-8C5E-B19544187533@freeswitch.org>

Well this isn't how the default config is to be used.  The external  
profile is for talking to things outside your organization like  
gateways and service providers.  Phones shouldn't be registering to  
them unless you bond the internal and external profiles together.


In the internal profile you have:

     <!-- used to share presence info across sofia profiles -->
     <!-- Name of the db to use for this profile -->
     <!--<param name="dbname" value="share_presence"/>-->
     <!--<param name="presence-hosts" value="$${domain}"/>-->
     <!-- ************************************************* -->


Then on external you have these settings:

     <!-- Name of the db to use for this profile -->
     <!--<param name="dbname" value="share_presence"/>-->
     <!--<param name="presence-hosts" value="$${domain}"/>-->
     <!--<param name="force-register-domain" value="$${domain}"/>-->
     <!--all inbound reg will stored in the db using this domain -->
     <!--<param name="force-register-db-domain" value="$${domain}"/>-->
     <!-- ************************************************* -->


The settings will allow you to bind two profiles together and make  
them act as one.   That would be what you want I suspect.

/b

On Dec 4, 2008, at 2:28 AM, Carole O. wrote:

> Since both were able to register I guess that the directory
> /directory/default/ is used by both sofia agent else user1002 should  
> not be
> able to register, isn't? I am confused here because I would have  
> said that
> the directory /directory/default/ was just for the users that  
> register on
> internal...
>
> I had a second problem: if user1001 tries to call user1002 it did  
> not work
> (but the opposite is ok). Has this something to do with the dialplan?



From anthony.minessale at gmail.com  Thu Dec  4 07:34:46 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Dec 2008 09:34:46 -0600
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <1228365981.25709.60.camel@develop4>
References: <1228352588.25709.42.camel@develop4>
	<191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
	<1228365981.25709.60.camel@develop4>
Message-ID: <191c3a030812040734s4f514f42s9a30a48c93709fd5@mail.gmail.com>

most likely it's because during the time you are dong artificial ringback
the other side is not doing RTP right.

When the call is answered we flush the rtp buffer and your missing audio is
probably flushed with it.
so you can choose to have a 3 second delay or erase the 3 seconds as it does
now.

This is a known problem with sonus which has been proven to build up an
audio delay during the time
you are waiting for the call to answer.  I'm sure you prefer the way it is
to a large audio delay.



On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <ack at telefonica.net>wrote:

> No TDM , all is SIP :
>
>
> PSTN ---> Sip Proxy_A --> FS ( brigde ) ringback/transfer_ringback
> -> Sip Proxy_B --> PSTN
>
>
> In logfile i think you can get some details about Media Gateways
> ( Sonus ) PSTN inbound / outbound is provided by Level3.
>
> I can get a capture of a call if you want, in capture the audio is not
> missing, issue with :
>
> - rtp buffer ?
> - Sonus ?
>
> Let me know anything you need so i can provide a log or create a new
> scenario.
>
>
> Thanks,
>
> El mi?, 03-12-2008 a las 22:12 -0600, Anthony Minessale escribi?:
> > what does PSTN represent?
> >
> > I know what the PSTN is but how are you reaching it?
> > is it TDM, SIP etc... what gateway type other details.
> >
> >
> > On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <ack at telefonica.net>
> > wrote:
> >         Hi guys,
> >
> >          I've a strange issue with FS , version svn -r10584 ,
> >         when FS bridges a call first 3 seconds of audio are missing ,
> >         looks that
> >         only happens on PSTN calls and using ringback or
> >         transfer_ringback. This
> >         only happens in calls from PSTN , not from VOIP. Some
> >         scenarios i tried
> >         to isolate this issue :
> >
> >
> >         - Issue
> >
> >         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
> >
> >         - Good setting bypass_media before run bridge but i need rtp
> >         in FS path
> >
> >         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
> >
> >         - Good
> >
> >         PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback ->
> >         PSTN
> >
> >         - Always good
> >
> >         VOIP --> FS ( brigde ) -> PSTN
> >
> >
> >         Dialplan has nothing wrong ( i guess ):
> >
> >         <extension name="Transfers">
> >            <condition field="destination_number"
> >         expression="^1??XXXXXXXXXX$">
> >              <action application="answer"/>
> >              <action application="speak" data="cepstral|Allison-8kHz|
> >         blah"/>
> >              <action application="set"
> >         data="hangup_after_bridge=false"/>
> >              <action application="set" data="playback_terminators=#"/>
> >              <action application="set" data="ringback=$${us-ring}"/>
> >              <action application="set" data="transfer_ringback=
> >         $${hold_music}"/>
> >              <action application="set" data="effective_caller_id_name=
> >         ${caller_id_name}"/>
> >              <action application="set"
> >         data="effective_caller_id_number=
> >         ${caller_id_number}"/>
> >              <action application="set" data="originate_timeout=30"/>
> >              <action application="set" data="call_timeout=30"/>
> >              <action application="bridge"
> >         data="sofia/default/18008226235 at PSTN_GW"/>
> >              <action application="speak" data="cepstral|Allison-8kHz|
> >         Transfer
> >         finished"/>
> >              <action application="hangup"/>
> >            </condition>
> >          </extension>
> >
> >
> >
> >         Any ideas ?
> >
> >         Attached log of FS ( F8 from console ).
> >
> >
> >         Thanks in advance !
> >
> >         --
> >         Angel Carpintero
> >         ack ( at ) telefonica ( dot ) net
> >
> >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
> >         6EF1 B90D
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
>
> --
> Angel Carpintero
> ack ( at ) telefonica ( dot ) net
>
> Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Dec  4 08:07:36 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 4 Dec 2008 08:07:36 -0800
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
Message-ID: <F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>

Gopal,

FreeSWITCH does not have free amd but you can buy a license. Please  
send a request to consulting at freeswitch.org. I have some experience  
with running amd so I can assist with setup questions.

-Michael

Sent from my iPhone

On Dec 4, 2008, at 4:50 AM, "Gopalakrishnan A.N" <saigop at gmail.com>  
wrote:

> Hi,
>
>    I would like to have predictive dialing. In asterisk we used  
> manager api and for outbound we use originate. The originate command  
> will dial a number where asterisk answer the call and then we  
> predict the answering machine with the silence file. Inspite of that  
> human voice is detected and transferred to agents.
>
>    Like this how can we go ahead with Freeswitch? Any help would be  
> appreciated.
>
> -- 
> Thank you  with regards,
> Gopal,
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gkuri at ieee.org  Thu Dec  4 09:25:56 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 04 Dec 2008 09:25:56 -0800
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <4937BD5C.7050002@gmail.com>
References: <4937BD5C.7050002@gmail.com>
Message-ID: <493812A4.4080404@ieee.org>

How about:

"The mailbox of the person you are trying to reach is full and can not
accept new messages at this time. Please try your call again later. Goodbye"

~Gabe

Tamas Cseke wrote:
> Hello,
> 
> I would like to use a disk quota in users' voicemail
> (http://jira.freeswitch.org/browse/MODAPP-173)
> 
> We haven't decided yet what would be the better prompts to play to the
> caller when the mailbox is full.
> Please advice some messages!
> 
> Thanks in advance,
> Tamas
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From msc at freeswitch.org  Thu Dec  4 09:43:26 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Dec 2008 09:43:26 -0800
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <493812A4.4080404@ieee.org>
References: <4937BD5C.7050002@gmail.com> <493812A4.4080404@ieee.org>
Message-ID: <87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>

Nice! I'll add that to my list of new prompts to be recorded. FYI, if
you have any other suggestions please email this list or post a
comment here:

http://jira.freeswitch.org/browse/FSSCRIPTS-9
-MC

On Thu, Dec 4, 2008 at 9:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
>
> How about:
>
> "The mailbox of the person you are trying to reach is full and can not
> accept new messages at this time. Please try your call again later. Goodbye"
>
> ~Gabe
>
> Tamas Cseke wrote:
> > Hello,
> >
> > I would like to use a disk quota in users' voicemail
> > (http://jira.freeswitch.org/browse/MODAPP-173)
> >
> > We haven't decided yet what would be the better prompts to play to the
> > caller when the mailbox is full.
> > Please advice some messages!
> >
> > Thanks in advance,
> > Tamas
> >
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Thu Dec  4 09:50:36 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Dec 2008 11:50:36 -0600
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>
References: <4937BD5C.7050002@gmail.com> <493812A4.4080404@ieee.org>
	<87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>
Message-ID: <191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>

maybe
http://www.sofaswitch.org/eg/sounds/fucked.wav



On Thu, Dec 4, 2008 at 11:43 AM, Michael Collins <msc at freeswitch.org> wrote:

> Nice! I'll add that to my list of new prompts to be recorded. FYI, if
> you have any other suggestions please email this list or post a
> comment here:
>
> http://jira.freeswitch.org/browse/FSSCRIPTS-9
> -MC
>
> On Thu, Dec 4, 2008 at 9:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >
> > How about:
> >
> > "The mailbox of the person you are trying to reach is full and can not
> > accept new messages at this time. Please try your call again later.
> Goodbye"
> >
> > ~Gabe
> >
> > Tamas Cseke wrote:
> > > Hello,
> > >
> > > I would like to use a disk quota in users' voicemail
> > > (http://jira.freeswitch.org/browse/MODAPP-173)
> > >
> > > We haven't decided yet what would be the better prompts to play to the
> > > caller when the mailbox is full.
> > > Please advice some messages!
> > >
> > > Thanks in advance,
> > > Tamas
> > >
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Dec  4 09:55:43 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Dec 2008 09:55:43 -0800
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>
References: <4937BD5C.7050002@gmail.com> <493812A4.4080404@ieee.org>
	<87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>
	<191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>
Message-ID: <87f2f3b90812040955h243bd649gdbca184c35815b96@mail.gmail.com>

I think GM Voices levies a "naughtiness surcharge" but I'll see what I
can find out. :)
-MC

On Thu, Dec 4, 2008 at 9:50 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> maybe
> http://www.sofaswitch.org/eg/sounds/fucked.wav
>
>
>
> On Thu, Dec 4, 2008 at 11:43 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> Nice! I'll add that to my list of new prompts to be recorded. FYI, if
>> you have any other suggestions please email this list or post a
>> comment here:
>>
>> http://jira.freeswitch.org/browse/FSSCRIPTS-9
>> -MC
>>
>> On Thu, Dec 4, 2008 at 9:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >
>> > How about:
>> >
>> > "The mailbox of the person you are trying to reach is full and can not
>> > accept new messages at this time. Please try your call again later.
>> > Goodbye"
>> >
>> > ~Gabe
>> >
>> > Tamas Cseke wrote:
>> > > Hello,
>> > >
>> > > I would like to use a disk quota in users' voicemail
>> > > (http://jira.freeswitch.org/browse/MODAPP-173)
>> > >
>> > > We haven't decided yet what would be the better prompts to play to the
>> > > caller when the mailbox is full.
>> > > Please advice some messages!
>> > >
>> > > Thanks in advance,
>> > > Tamas
>> > >
>> > >
>> > >
>> > > _______________________________________________
>> > > Freeswitch-users mailing list
>> > > Freeswitch-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Thu Dec  4 10:03:20 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2008 12:03:20 -0600
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <87f2f3b90812040955h243bd649gdbca184c35815b96@mail.gmail.com>
References: <4937BD5C.7050002@gmail.com> <493812A4.4080404@ieee.org>
	<87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>
	<191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>
	<87f2f3b90812040955h243bd649gdbca184c35815b96@mail.gmail.com>
Message-ID: <076D0E41-DE7A-494C-8096-352BBE104246@freeswitch.org>

Actually GM Voices won't do anything with profanity in it.

/b

On Dec 4, 2008, at 11:55 AM, Michael Collins wrote:

> I think GM Voices levies a "naughtiness surcharge" but I'll see what I
> can find out. :)
> -MC



From odermann at googlemail.com  Thu Dec  4 11:45:36 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 4 Dec 2008 20:45:36 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
Message-ID: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>

hi,

after we managed to setup fs with mod_fax and our socket outbound
script, we have some questioons about an error and problems, when
sending a fax:

1.) there is one error, we get always - no matter, if the fax was sent
successfully or not.
in the pastebin under http://pastebin.freeswitch.org/6338 you can see
the error in the last line.
this is the full log of a fax in fs console loglevel debug.


2.) fax works quite good. we couls send long faxes over a normal fax
machine without any problem.
but for fast testing we are using a softfax (fritz fax). here we have
some more problems.
mostly a fax with one page will pass through, but more pages will mostly fail.
because we are new to the fax thing, we do not really know, what the
messages tell us about failed faxes.
here are the top 3 messages (unordered and always one at a time -
nerver at once):

variable_fax_result_text => Received a DCN while waiting for a DIS

fax_result_text => The HDLC carrier did not stop in a timely manner

fax_result_text => Unexpected message received


could someone please tell us, where the problem might be?


thanks
dennis


From msc at freeswitch.org  Thu Dec  4 12:09:21 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Dec 2008 12:09:21 -0800
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
Message-ID: <87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>

Dennis,

Thanks for your input on the fax stuff! We will check this out and report back.

Question: if one of the devs would like to SSH into your system to do
further testing, is that okay?

Thanks,
MC

On Thu, Dec 4, 2008 at 11:45 AM, Dennis <odermann at googlemail.com> wrote:
> hi,
>
> after we managed to setup fs with mod_fax and our socket outbound
> script, we have some questioons about an error and problems, when
> sending a fax:
>
> 1.) there is one error, we get always - no matter, if the fax was sent
> successfully or not.
> in the pastebin under http://pastebin.freeswitch.org/6338 you can see
> the error in the last line.
> this is the full log of a fax in fs console loglevel debug.
>
>
> 2.) fax works quite good. we couls send long faxes over a normal fax
> machine without any problem.
> but for fast testing we are using a softfax (fritz fax). here we have
> some more problems.
> mostly a fax with one page will pass through, but more pages will mostly fail.
> because we are new to the fax thing, we do not really know, what the
> messages tell us about failed faxes.
> here are the top 3 messages (unordered and always one at a time -
> nerver at once):
>
> variable_fax_result_text => Received a DCN while waiting for a DIS
>
> fax_result_text => The HDLC carrier did not stop in a timely manner
>
> fax_result_text => Unexpected message received
>
>
> could someone please tell us, where the problem might be?
>
>
> thanks
> dennis
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Dec  4 12:17:40 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2008 14:17:40 -0600
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
Message-ID: <195B4EA0-8548-4FA7-8B72-DE2315D69F7B@freeswitch.org>

Also need to know is this via SIP or TDM?

/b

On Dec 4, 2008, at 2:09 PM, Michael Collins wrote:

> Dennis,
>
> Thanks for your input on the fax stuff! We will check this out and  
> report back.
>
> Question: if one of the devs would like to SSH into your system to do
> further testing, is that okay?
>
> Thanks,
> MC



From brian at freeswitch.org  Thu Dec  4 12:17:40 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2008 14:17:40 -0600
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
Message-ID: <195B4EA0-8548-4FA7-8B72-DE2315D69F7B@freeswitch.org>

Also need to know is this via SIP or TDM?

/b

On Dec 4, 2008, at 2:09 PM, Michael Collins wrote:

> Dennis,
>
> Thanks for your input on the fax stuff! We will check this out and  
> report back.
>
> Question: if one of the devs would like to SSH into your system to do
> further testing, is that okay?
>
> Thanks,
> MC



From odermann at googlemail.com  Thu Dec  4 12:25:14 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 4 Dec 2008 21:25:14 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
Message-ID: <5e414ed0812041225x4efa682ev17b1a366ecfd5654@mail.gmail.com>

my server (including me) is your slave. anthony already feels home on
my server, so you are greatly invited ;-)

it is quite late in germany, so i feel, that we should meet in irc
tomorrow, if this is ok for you.


@brian
we get everthing over sip. so we receive the faxes over sip. the
faxes, which we send for testing (not over fs or the same machine) are
sent over isdn.


2008/12/4 Michael Collins <msc at freeswitch.org>:
> Dennis,
>
> Thanks for your input on the fax stuff! We will check this out and report back.
>
> Question: if one of the devs would like to SSH into your system to do
> further testing, is that okay?


From steveu at coppice.org  Thu Dec  4 16:23:33 2008
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 05 Dec 2008 08:23:33 +0800
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
Message-ID: <49387485.9000303@coppice.org>

Dennis wrote:
> hi,
>
> after we managed to setup fs with mod_fax and our socket outbound
> script, we have some questioons about an error and problems, when
> sending a fax:
>
> 1.) there is one error, we get always - no matter, if the fax was sent
> successfully or not.
> in the pastebin under http://pastebin.freeswitch.org/6338 you can see
> the error in the last line.
> this is the full log of a fax in fs console loglevel debug.
>   
That looks like something annoying but harmless. It looks like the comms 
path is disabled slightly before the flow of packets is turned off. 
That's probably just a silly slip in the code.
>
> 2.) fax works quite good. we couls send long faxes over a normal fax
> machine without any problem.
> but for fast testing we are using a softfax (fritz fax). here we have
> some more problems.
> mostly a fax with one page will pass through, but more pages will mostly fail.
> because we are new to the fax thing, we do not really know, what the
> messages tell us about failed faxes.
> here are the top 3 messages (unordered and always one at a time -
> nerver at once):
>
> variable_fax_result_text => Received a DCN while waiting for a DIS
>
> fax_result_text => The HDLC carrier did not stop in a timely manner
>
> fax_result_text => Unexpected message received
>
>
> could someone please tell us, where the problem might be?
>   
Does Fritz FAX means the ISDN card stuff? If so, that should be 
something well proven. However, the errors you are getting sound like 
the FAX at the far end is buggy. I think a log of the audio from one or 
two of these calls is needed for analysis.

Regards,
Steve



From dave at 3c.co.uk  Fri Dec  5 00:15:37 2008
From: dave at 3c.co.uk (David Knell)
Date: Fri, 05 Dec 2008 08:15:37 +0000
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <076D0E41-DE7A-494C-8096-352BBE104246@freeswitch.org>
References: <4937BD5C.7050002@gmail.com>
	<493812A4.4080404@ieee.org>	<87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>	<191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>	<87f2f3b90812040955h243bd649gdbca184c35815b96@mail.gmail.com>
	<076D0E41-DE7A-494C-8096-352BBE104246@freeswitch.org>
Message-ID: <4938E329.9030908@3c.co.uk>

I still know some folk in the 900-number business.  They won't do 
anything without
profanity in it ;-)

--Dave
> Actually GM Voices won't do anything with profanity in it.
>
> /b
>
> On Dec 4, 2008, at 11:55 AM, Michael Collins wrote:
>
>   
>> I think GM Voices levies a "naughtiness surcharge" but I'll see what I
>> can find out. :)
>> -MC
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From faisalmaqsoodi at yahoo.com  Fri Dec  5 00:37:24 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Fri, 5 Dec 2008 00:37:24 -0800 (PST)
Subject: [Freeswitch-users] Handling directory of sound files
Message-ID: <228017.80650.qm@web30704.mail.mud.yahoo.com>

Hi,
?Can i accomplish folder tasks with freeswitch? For instance, i need to play all sound files contained in a directory sequentially or randomly. Plz help me doing this.
????????????????????????????????????????????????????????????????????????????????????????????????? Faisal



      
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From msc at freeswitch.org  Fri Dec  5 00:36:09 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 00:36:09 -0800
Subject: [Freeswitch-users] Nice FS article
Message-ID: <87f2f3b90812050036n72d85fbcqf24a2d1f0c878a8d@mail.gmail.com>

Check it out:
http://digg.com/software/FreeSWITCH_knocks_Asterisk_s_block_off

Please diggit left and right!!

-MC


From msc at freeswitch.org  Fri Dec  5 00:47:04 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 00:47:04 -0800
Subject: [Freeswitch-users] Handling directory of sound files
In-Reply-To: <228017.80650.qm@web30704.mail.mud.yahoo.com>
References: <228017.80650.qm@web30704.mail.mud.yahoo.com>
Message-ID: <87f2f3b90812050047i73e34e3u10f9c1b4b1fb4704@mail.gmail.com>

Is this for Music on Hold? Or is it for a different application altogether?
Thanks,
MC

On Fri, Dec 5, 2008 at 12:37 AM, Faisal Maqsoodi
<faisalmaqsoodi at yahoo.com> wrote:
> Hi,
>  Can i accomplish folder tasks with freeswitch? For instance, i need to play
> all sound files contained in a directory sequentially or randomly. Plz help
> me doing this.
>
> Faisal
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From faisalmaqsoodi at yahoo.com  Fri Dec  5 00:59:48 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Fri, 5 Dec 2008 00:59:48 -0800 (PST)
Subject: [Freeswitch-users] Handling directory of sound files
In-Reply-To: <87f2f3b90812050047i73e34e3u10f9c1b4b1fb4704@mail.gmail.com>
Message-ID: <692105.73475.qm@web30706.mail.mud.yahoo.com>

Its not without music on hold completely. Say, e.g, moh is being played but when i press 1 it should start playing files contained in a specific directory sequentially or randomly. Hope i m able to explain.
? ? ? ? ? ? ? ? ? ? ? ?? ???????????????????????? Faisal

--- On Fri, 12/5/08, Michael Collins <msc at freeswitch.org> wrote:
From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Handling directory of sound files
To: freeswitch-users at lists.freeswitch.org
Date: Friday, December 5, 2008, 12:47 AM

Is this for Music on Hold? Or is it for a different application altogether?
Thanks,
MC

On Fri, Dec 5, 2008 at 12:37 AM, Faisal Maqsoodi
<faisalmaqsoodi at yahoo.com> wrote:
> Hi,
>  Can i accomplish folder tasks with freeswitch? For instance, i need to
play
> all sound files contained in a directory sequentially or randomly. Plz
help
> me doing this.
>
> Faisal
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From mrjoebain at gmail.com  Fri Dec  5 01:10:23 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Fri, 5 Dec 2008 09:10:23 +0000
Subject: [Freeswitch-users] Javascript ODBC on Windows
In-Reply-To: <E05E9383-0BF2-4B8B-93A5-09FB99817F19@jerris.com>
References: <748d53500812020629p6a0d178dh672cec871c018254@mail.gmail.com>
	<E05E9383-0BF2-4B8B-93A5-09FB99817F19@jerris.com>
Message-ID: <748d53500812050110u6fffe498tb3fe6ff5b64600af@mail.gmail.com>

Thanks, you're right it seems to be an odbc problem, 64bit / 32bit clash I
think.

Joe

2008/12/2 Michael Jerris <mike at jerris.com>

> Yes, it should work fine.  As the error message says it didn't find
> the data source name you specified.  You need to setup your odbc data
> source on the system
>
> Mike
>
> On Dec 2, 2008, at 9:29 AM, Joe Bain wrote:
>
> > Hi all,
> >
> > Is it possible to use mod_spidermonkey_odbc with a Windows
> > installation of FreeSWITCH at the moment? If so does anyone have any
> > pointers? I get:
> >
> > 2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145
> > switch_odbc_handle_connect() Connecting ivr_test
> > 2008-12-02 14:23:57 [ERR] switch_odbc.c:160
> > switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [Microsoft]
> > [ODBC Driver Manager] Data source name not found and no default
> > driver specified
> >
> > when I try.
> >
> > Thanks in advance,
> >
> > Joe Bain
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mrjoebain at gmail.com  Fri Dec  5 02:33:42 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Fri, 5 Dec 2008 10:33:42 +0000
Subject: [Freeswitch-users] Problem reloading xml
Message-ID: <748d53500812050233y3b965e13ofcb566a1e83bbb14@mail.gmail.com>

Hi all,

I have come across a strange problem when using the phrases in conf/lang/en.
Initially I had a problem where FreeSwitch wouldn't load new subdirectories,
even when I included their paths in the en.xml file. I went ahead writing
all the phrases (I only have 3 so far) in en.xml but now it doesn't
recognise even these properly. I thought there might be some conflicts with
the naming of phrases so I deleted all the included demo phrases and
commented their lines out of the en.xml file but when I do 'reloadxml' I get
the error:

Error including C:\Program Files (x86)\FreeSWITCH\conf\lang\en\demo/*.xml
(No su
ch file or directory)
Error including C:\Program Files (x86)\FreeSWITCH\conf\lang\en\test/*.xml
(No su
ch file or directory)
Error including C:\Program Files (x86)\FreeSWITCH\conf\lang\en\vm/sounds.xml
(No
 such file or directory)

This persists after restarting FS too. Very confusing, any ideas? I am
running FS on Vista 64 bit, but with the 32 bit version. The version of FS
is "FreeSWITCH Version 1.0.trunk (10175M)".

Thanks

Joe
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From odermann at googlemail.com  Fri Dec  5 02:54:02 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 5 Dec 2008 11:54:02 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <49387485.9000303@coppice.org>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
Message-ID: <5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>

2008/12/5 Steve Underwood <steveu at coppice.org>:
>> 1.) there is one error, we get always - no matter, if the fax was sent
>> successfully or not.
>> in the pastebin under http://pastebin.freeswitch.org/6338 you can see
>> the error in the last line.
>> this is the full log of a fax in fs console loglevel debug.
>>
> That looks like something annoying but harmless. It looks like the comms
> path is disabled slightly before the flow of packets is turned off.
> That's probably just a silly slip in the code.

yup, because the error always appear, i successful or not, this error
can't be a big problem. i just do not like red lines in the log ;-)


> Does Fritz FAX means the ISDN card stuff? If so, that should be
> something well proven. However, the errors you are getting sound like
> the FAX at the far end is buggy. I think a log of the audio from one or
> two of these calls is needed for analysis.

yes, fritz fax is the isdn stuff. normally it works very well.
how can i get a log of the audio? when a fax is coming in, there
happens quite little in the console at loglevel debug. i pasted all
into the pastebin.
if there are more possibilities to get mor information, please let me know.


From jan.kubr at gmail.com  Fri Dec  5 03:08:24 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Fri, 5 Dec 2008 12:08:24 +0100
Subject: [Freeswitch-users] DTMF from cell phones
Message-ID: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>

Hi,
recently someone was mentioning an issue with DTMF here, but there was
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
(read app doesn't terminate). I could not find any regularity in this,
sometimes it is recognized just fine, sometimes I had to wait for the
file to be played etc. The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is
recognized perfectly. So it's probably related to GSM or something.

I was wondering whether anyone experienced the same and whether there
is something I can do about it. There are a few DTMF-related variables
in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
them a bit, but I don't really know what I'm doing.. Couldn't find any
docs, either.
Any ideas would be appreciated.

Jan Kubr


From saigop at gmail.com  Fri Dec  5 03:23:36 2008
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Fri, 5 Dec 2008 16:53:36 +0530
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
	<F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
Message-ID: <2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>

Hi Micheal,
  Thanks for the reply! cant I try with tone detect?

  Like dial a number in session and try to detect with tone detect and then
bridge the call with some extension.

-- 
Thank you  with regards,
Gopal,
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From faisalmaqsoodi at yahoo.com  Fri Dec  5 03:48:13 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Fri, 5 Dec 2008 03:48:13 -0800 (PST)
Subject: [Freeswitch-users] Handling directory of sound files
Message-ID: <132465.82190.qm@web30706.mail.mud.yahoo.com>

Its not without music on hold completely. Say, e.g, moh is being played
but when i press 1 it should start playing files contained in a
specific directory sequentially or randomly. I havent got any solution to this problem yet. Can anyone plz guide me to some documentation or anything else regarding this matter.
? ? ? ? ? ? ? ? ? ? ? ?? ???????????????????? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? Faisal


      
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From Prometheus001 at gmx.net  Fri Dec  5 03:54:54 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Dec 2008 12:54:54 +0100
Subject: [Freeswitch-users] Event_Socket: How to determine the ogininater
 uuid before an outgoing call is set up?
Message-ID: <4939168E.9020400@gmx.net>

I am building an IVR application where an incoming call is initiating an
outgoing call. When I pass a "variable_other_uuid" (the uuid of the
incoming channel) at originate time, I am able to reference to the
incomig call, once the outgoing call is set up. So the outgoing call can
see the uuid of the incoming call who has originated the outgoing call.
This is needed for bridging the 2 calls together.

However I want to control also the call setup process (see, if the
outgoing call is ringing etc.). At call setup time, when I parse the
channel_originate ,channel_outgoing and channel_progress events, I
cannot see any reference to the incoming call (variable_other_uuid is
not set). I suspect that variables are only passed once the outgoing
channel is set up.

Has anybody an idea, how I may get the uuid of the originating uuid in
the outgoing call at call setup?

Best regards
Peter


From regs at kinetix.gr  Fri Dec  5 04:23:28 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 14:23:28 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
Message-ID: <49391D40.6050103@kinetix.gr>

The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this nornal?

And another question : what piece of info could inform me about who 
hanged up?


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 



From ack at telefonica.net  Fri Dec  5 04:30:25 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Fri, 05 Dec 2008 13:30:25 +0100
Subject: [Freeswitch-users] DTMF from cell phones
In-Reply-To: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>
References: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>
Message-ID: <1228480225.25709.100.camel@develop4>

I had some issues with some previous versions of FS , in trunk looks
that is fixed. ( Notice current svn revision is 10609 )

in sip profiles i have :

...
<param name="rfc2833-pt" value="101"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${global_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="none"/>
<param name="inbound-codec-negotiation" value="greedy"/>
...

As codecs g711 ULAW (PCMU):

in vars.xml.conf :

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM"/>

So i guess that using latest version with a few changes in your config 
should work unless there's any other issue related to your sip provider
( PSTN / Media Gateway ), on this case you can get some captures of
sip/rtp traffic to check SDP and rtp Marks.

El vie, 05-12-2008 a las 12:08 +0100, Jan Kubr escribi?:
> Hi,
> recently someone was mentioning an issue with DTMF here, but there was
> no solution. I have a similar problem, when calling Freeswitch from my
> cell phone (via a SIP provider), sometimes DTMF is not recognized
> (read app doesn't terminate). I could not find any regularity in this,
> sometimes it is recognized just fine, sometimes I had to wait for the
> file to be played etc. The important thing to note is that when using
> a SIP softphone (X-Lite) I have never had this problem, DTMF is
> recognized perfectly. So it's probably related to GSM or something.
> 
> I was wondering whether anyone experienced the same and whether there
> is something I can do about it. There are a few DTMF-related variables
> in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
> dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
> them a bit, but I don't really know what I'm doing.. Couldn't find any
> docs, either.
> Any ideas would be appreciated.
> 
> Jan Kubr
> 

Cheers,
-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

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From mike at jerris.com  Fri Dec  5 06:39:14 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 09:39:14 -0500
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
Message-ID: <3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>


On Dec 5, 2008, at 5:54 AM, Dennis wrote:

> 2008/12/5 Steve Underwood <steveu at coppice.org>:
>>> 1.) there is one error, we get always - no matter, if the fax was  
>>> sent
>>> successfully or not.
>>> in the pastebin under http://pastebin.freeswitch.org/6338 you can  
>>> see
>>> the error in the last line.
>>> this is the full log of a fax in fs console loglevel debug.
>>>
>> That looks like something annoying but harmless. It looks like the  
>> comms
>> path is disabled slightly before the flow of packets is turned off.
>> That's probably just a silly slip in the code.
>
> yup, because the error always appear, i successful or not, this error
> can't be a big problem. i just do not like red lines in the log ;-)
>
>
>> Does Fritz FAX means the ISDN card stuff? If so, that should be
>> something well proven. However, the errors you are getting sound like
>> the FAX at the far end is buggy. I think a log of the audio from  
>> one or
>> two of these calls is needed for analysis.
>
> yes, fritz fax is the isdn stuff. normally it works very well.
> how can i get a log of the audio? when a fax is coming in, there
> happens quite little in the console at loglevel debug. i pasted all
> into the pastebin.
> if there are more possibilities to get mor information, please let  
> me know.

You should be able to record session with stereo to get the audio  
here, the other easy way would be to use pcapsipdump tool (look for it  
on the wiki) to get a pcap of the whole call.

Mike



From mike at jerris.com  Fri Dec  5 06:40:55 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 09:40:55 -0500
Subject: [Freeswitch-users] DTMF from cell phones
In-Reply-To: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>
References: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>
Message-ID: <58C18F16-E8A3-4963-A624-32BFE13D2C26@jerris.com>


On Dec 5, 2008, at 6:08 AM, Jan Kubr wrote:

> Hi,
> recently someone was mentioning an issue with DTMF here, but there was
> no solution. I have a similar problem, when calling Freeswitch from my
> cell phone (via a SIP provider), sometimes DTMF is not recognized
> (read app doesn't terminate). I could not find any regularity in this,
> sometimes it is recognized just fine, sometimes I had to wait for the
> file to be played etc. The important thing to note is that when using
> a SIP softphone (X-Lite) I have never had this problem, DTMF is
> recognized perfectly. So it's probably related to GSM or something.
>
> I was wondering whether anyone experienced the same and whether there
> is something I can do about it. There are a few DTMF-related variables
> in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
> dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
> them a bit, but I don't really know what I'm doing.. Couldn't find any
> docs, either.
> Any ideas would be appreciated.

If it is coming from the sip provider as rfc 2833 dtmf, they are  
probably doing inband detection and failing at it.  If you look at an  
rtp dump you can confirm this.  If this is the case, there is nothing  
you can do on the FreeSWITCH side and the provider will have to fix it.

Mike


From mike at jerris.com  Fri Dec  5 06:41:29 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 09:41:29 -0500
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
	<F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
	<2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>
Message-ID: <DB682CA7-D304-4D78-ABA7-4BD91FF6A8AA@jerris.com>


On Dec 5, 2008, at 6:23 AM, Gopalakrishnan A.N wrote:

> Hi Micheal,
>
>   Thanks for the reply! cant I try with tone detect?
>
>   Like dial a number in session and try to detect with tone detect  
> and then bridge the call with some extension.

If you know the exact frequency of the tone you can, but I suspect you  
do not.

Mike



From mike at jerris.com  Fri Dec  5 06:42:38 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 09:42:38 -0500
Subject: [Freeswitch-users] Event_Socket: How to determine the
	ogininater uuid before an outgoing call is set up?
In-Reply-To: <4939168E.9020400@gmx.net>
References: <4939168E.9020400@gmx.net>
Message-ID: <41CC850A-05A4-4618-B413-A655A7CCEA38@jerris.com>


On Dec 5, 2008, at 6:54 AM, Peter P GMX wrote:

> I am building an IVR application where an incoming call is  
> initiating an
> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
> incoming channel) at originate time, I am able to reference to the
> incomig call, once the outgoing call is set up. So the outgoing call  
> can
> see the uuid of the incoming call who has originated the outgoing  
> call.
> This is needed for bridging the 2 calls together.
>
> However I want to control also the call setup process (see, if the
> outgoing call is ringing etc.). At call setup time, when I parse the
> channel_originate ,channel_outgoing and channel_progress events, I
> cannot see any reference to the incoming call (variable_other_uuid is
> not set). I suspect that variables are only passed once the outgoing
> channel is set up.

Control in what way?

>
>
> Has anybody an idea, how I may get the uuid of the originating uuid in
> the outgoing call at call setup?



From zolotov at altron.ua  Fri Dec  5 06:53:05 2008
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Fri, 05 Dec 2008 16:53:05 +0200
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
Message-ID: <1228488785.5481.4.camel@opos20.altron.lan>

  Greetings!  

Question about possibility of the use FreeSWITCH for work with T1/E1
streams under Sun Solaris 10 a bit clears up (Solaris 11 is in condition
of alpha-version and not suitable for the industrial use). But answers
carry more negative sense.

Start of T1/E1 under Sun Solaris has 2 stages: a) start of wanpipe's
interface; b) make FreeSWITCH for Sun Solaris.


a) For Linux Sangoma recommends
(http://wiki.sangoma.com/wanpipe-freeswitch-install) installation of new
interface wanpipe-3.3.14.tgz, which has beta-status and placed at
ftp://ftp.sangoma.com/linux/current_wanpipe.

This installation and tests were successful.

In this release (2007 y.) by Sangoma was added TDM API - native
interface for FreeSwitch (YATE) ? which is absent in all previous
releases of WANPIPE. In the Linux release 3.3.14 this variant was named
as ?TDM API?, in older releases - ?TDM Voice?.

In this case FreeSWITCH works very good with E1/T1 cards
A-101/102/104/108 without installation in zaptel system, and spans in
configuration files of FreeSWITCH declared as [wanpipe#].

But for Sun Solaris we found only
drverftp://ftp.sangoma.com/Solaris/Beta/SVwanpipe-i386-5.10.pkgbeta,
which dated 2007:

> NAME=sangoma.com Wanpipe Driver

> VERSION=1.1.0,REV=2007-07-16,

Packet Svwanpipe-i386-5.10.pkg uses Svzaptel-i386-5.10.pkg, which wasn't
developed by Sangoma, but by little free
communityhttp://www.solarisvoip.com/, source codes of this packet is
herehttps://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk/

The analysis of source codes shows that this project develops very
slowly (there are no updates for about year), has very limited
functionality and supports ( unlike original zaptel ) very limited list
of cards ( only one ;) - Digium Wildcard TE110P T1/PRI).

Svwanpipe-i386-5.10.pkg supports only 64-bit Sun Solaris (on CD, which
we get with Sangoma's cards, presents 32-bit driver and PDF document
about installation under Sun Solaris ? but it dated 2001 ? 2002 yy).

At first we have checked up installation with TDM Voice + zaptel (like
for Asterisk) under Linux. We configured PRI spans as [zt] ... - such
installation works good and we could do calls :


> originate openzap/1/A/20000 &sleep(3)


      * evidently that call retranslates from span 1 to span 2,
        connected with cross-cable, and goes to extension 20000.
        


b) About making FreeSwitch:

Under Sun Solaris 10 with GCC makes FreeSWITCH core and most modules,
except openzapand some others, because Sun Solaris 10 has GCC 3.4,
but ./configure for openzap requires compatibility with ANSII 99.

We have checked 3 different methods of makinf FreeSWITCH :


1. set up GCC 4.0.2 from CSW-repositaries (and all reguired for GCC
*.pkg);

2. set up SunStudio 12 and do make cc/CC; 

3.cross-compile FreeSwitch for Sun Solaris under Linux with CC-options.


In testing we used assembly SunStudio as 32-bit application (64-bit
comes to the end with mistakes of assembly of some libraries, it is
possible to correct for it easily, but we did not begin to specify it).


For testing cards A-104 we have repeated the same installation and
configuring as under Linux (wanpipe TDM Voice + zaptel) on 4 different
64-bit servers under Sun Solaris 10:


      * on 3 servers (2 of them manufactured by SUN) executed
        successfully : # wanrouter start
        


4 wanrouter# interfaces was created; leds on spans, connected with
cross-cable, becomes GREEN, i.e. synchronization T1/E1 presents (no
alarms).

On these servers FreeSWITCH correctly makes:


> load mod_openzap


      * but when we make :
        


> originate openzap/1/A/20000 &sleep(3)


- for 2 connected with crosss-cable spans (1 & 2) FreeSWITCH transmits
PRI message (chan 1/31), but chan 2/31 doesn't receive this message
(unlike under Linux) and call breaks after timeout.

On 4-th server wanpipe doesn't even starts by mistake some IOCTL (i.e.
at a command for device). Three ?working? servers is on AMD Opteron Dual
Core 2214(F), fourth is on ntel Xeon 3210.

So, we supposes Svwanpipe-i386-5.10.pkg or Svzaptel-i386-5.10.pkg from
Sangoma checked up a little and on some one processor and doesn't heave
up even on beta, as they are declared, at the best on alpha


What's the reason of error???

      * wanpipe doesn't work under Solaris?
        
      * wrong working signalling with zaptel? (but same configuration
        works good under Linux)
        
      * wrong working of FreeSWITCH, which was built correctly, but
        their work was violated?
      * 
      * 
Thanks, Evgeniy.





From msc at freeswitch.org  Fri Dec  5 07:08:18 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 07:08:18 -0800
Subject: [Freeswitch-users] Handling directory of sound files
Message-ID: <B25E3A72-98AE-4013-AAE0-CE978FD28717@freeswitch.org>

Check out mod_localstream on the wiki and see if that sounds like what  
you need. I'm still learning it all myself but I believe that's where  
you should start. Please report back with any questions and we will  
take it from there!

-MC


On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>  
wrote:

> Its not without music on hold completely. Say, e.g, moh is being  
> played but when i press 1 it should start playing files contained in  
> a specific directory sequentially or randomly. I havent got any  
> solution to this problem yet. Can anyone plz guide me to some  
> documentation or anything else regarding this matter.
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>                                                                      
> Faisal
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frank at impactfax.com  Fri Dec  5 07:22:13 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 10:22:13 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <20804652.post@talk.nabble.com>
Message-ID: <1fdd01c956ed$4003d280$33014c0a@ws4>


Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone (a key press, say 2) and when FS
hears that tone, then FS can broadcast another key tone (say 6) back to
the channels?




From pmhshz at gmail.com  Fri Dec  5 07:31:01 2008
From: pmhshz at gmail.com (shehzad p)
Date: Fri, 5 Dec 2008 07:31:01 -0800 (PST)
Subject: [Freeswitch-users] How to setup TLS between two Freeswitch servers
Message-ID: <20856369.post@talk.nabble.com>



I am wondering how to setup two freeswitch servers to route call with TLS
configured between them.

As shown in wiki http://wiki.freeswitch.org/wiki/SIP_TLS, I created two
certificates on one freeswitch, and changed SIP profile by enabling tls in
it,
then Starting freeswitch it just opens port 5061 (for TLS ),

But when i route the call from that FS server, it uses the its general ports
(5060 and 5080) for call.

Where i am missing something?, 
A doubt is about where to place which certificate. 
-- 
View this message in context: http://www.nabble.com/How-to-setup-TLS-between-two-Freeswitch-servers-tp20856369p20856369.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From carole.olivier at enst.fr  Fri Dec  5 07:35:51 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Fri, 5 Dec 2008 07:35:51 -0800 (PST)
Subject: [Freeswitch-users] conference configured to call automatically the
 attended does not work
Message-ID: <20856465.post@talk.nabble.com>


Hello,

I have got some problems for the configuration of a simple conference which
should be established by calling an extension and automatically inviting 2
people.

Actually, this is based on the default configuration of Freeswitch
(extension 0911). I have changed it a little:

<extension name="conference_test">
    <condition field="destination_number" expression="^0911$">
    <!--  These parameters effect the outcalls made once you join 
      --> 
      <action application="set"
data="conference_auto_outcall_caller_id_name=telephoneX" /> 
      <action application="set"
data="conference_auto_outcall_id_number=0911" /> 
      <action application="set" data="conference_auto_outcall_timeout=60" /> 
      <action application="set" data="conference_auto_outcall_flags=none" /> 
 
    <!--  list of devices called  --> 
      <action application="conference_set_auto_outcall"
data="sofia/internal/1010@$${domain}" /> 
      <action application="conference_set_auto_outcall"
data="sofia/internal/1002@$${domain}" /> 

      <action application="conference" data="conferenceTest at default" /> 
    </condition>
 </extension>


I have attached a file with the console errors. There are some errors (moh
errors) but since these were also present for room conference and it did not
prevent it for working, I guess this is not the fundamental reason for the
previous problem.

I have an additional question. I have installed freeswitch from
opensuse.org, there is a simple "one-click installation" but I am not sure
this was a good idea, it seems to be light isn't?

Thanks for your help,
Carole
http://www.nabble.com/file/p20856465/error_console.txt error_console.txt 
-- 
View this message in context: http://www.nabble.com/conference-configured-to-call-automatically-the-attended-does-not-work-tp20856465p20856465.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Fri Dec  5 07:36:19 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 09:36:19 -0600
Subject: [Freeswitch-users] How to setup TLS between two Freeswitch
	servers
In-Reply-To: <20856369.post@talk.nabble.com>
References: <20856369.post@talk.nabble.com>
Message-ID: <7D08D1D5-5FA8-40FD-BF82-EA9412F6E0D2@freeswitch.org>

You would use something like this sofia/profile/ 
user at remotefsip;transport=tls

/b

On Dec 5, 2008, at 9:31 AM, shehzad p wrote:

>
>
> I am wondering how to setup two freeswitch servers to route call  
> with TLS
> configured between them.



From msc at freeswitch.org  Fri Dec  5 07:44:30 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 07:44:30 -0800
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <1fdd01c956ed$4003d280$33014c0a@ws4>
References: <1fdd01c956ed$4003d280$33014c0a@ws4>
Message-ID: <C7F15A37-CCA9-4CAE-AA30-C012224BFBC6@freeswitch.org>

What would need to happen after the tone is sent back out? Also, would  
this be part of something like an IVR?

-MC


On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

>
> Is there any dialplan instructions that could be added that would sit
> and listen during a call for a tone (a key press, say 2) and when FS
> hears that tone, then FS can broadcast another key tone (say 6) back  
> to
> the channels?
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From frank at impactfax.com  Fri Dec  5 07:51:07 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 10:51:07 -0500
Subject: [Freeswitch-users] key tone trigger event during call
Message-ID: <201201c956f1$49abe940$33014c0a@ws4>

Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone (a key press, say 2) and when FS
hears that tone, then FS can broadcast another key tone (say 6) back to
the channels?
 
-Frank
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From msc at freeswitch.org  Fri Dec  5 07:56:32 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 07:56:32 -0800
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <49391D40.6050103@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
Message-ID: <70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>

I will do some research on this and let you know what I find out.  
Question: are these internal calls or pstn or ?? Just curious about  
your environment.

Thanks,
MC



On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>  
wrote:

> The proto_specific_hangup_cause is missing on one of the two
> call legs. When the caller hangs up it is missing from the a-leg CDR.
> When the callee hangs up it is missing from the b-leg CDR. Is this  
> nornal?
>
> And another question : what piece of info could inform me about who
> hanged up?
>
>
> -- 
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From frank at impactfax.com  Fri Dec  5 08:00:17 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 11:00:17 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <C7F15A37-CCA9-4CAE-AA30-C012224BFBC6@freeswitch.org>
Message-ID: <203101c956f2$91781450$33014c0a@ws4>

After the tone is sent back out, we are done.  There is nothing left to
do.  
No, this key press detection is during a bridged call between two
parties.  No IVR here.   So, FS hears a key press tone during a call and
then responds to the parties with another/different key press tone.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
 
What would need to happen after the tone is sent back out? Also, would  
this be part of something like an IVR?

-MC


On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

>
> Is there any dialplan instructions that could be added that would sit
> and listen during a call for a tone (a key press, say 2) and when FS
> hears that tone, then FS can broadcast another key tone (say 6) back  
> to
> the channels?
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From msc at freeswitch.org  Fri Dec  5 08:02:35 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 08:02:35 -0800
Subject: [Freeswitch-users] Event_Socket: How to determine the
	ogininater uuid before an outgoing call is set up?
In-Reply-To: <4939168E.9020400@gmx.net>
References: <4939168E.9020400@gmx.net>
Message-ID: <F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>

What is your originate string?
-MC


On Dec 5, 2008, at 3:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> I am building an IVR application where an incoming call is  
> initiating an
> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
> incoming channel) at originate time, I am able to reference to the
> incomig call, once the outgoing call is set up. So the outgoing call  
> can
> see the uuid of the incoming call who has originated the outgoing  
> call.
> This is needed for bridging the 2 calls together.
>
> However I want to control also the call setup process (see, if the
> outgoing call is ringing etc.). At call setup time, when I parse the
> channel_originate ,channel_outgoing and channel_progress events, I
> cannot see any reference to the incoming call (variable_other_uuid is
> not set). I suspect that variables are only passed once the outgoing
> channel is set up.
>
> Has anybody an idea, how I may get the uuid of the originating uuid in
> the outgoing call at call setup?
>
> Best regards
> Peter
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Fri Dec  5 08:07:42 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 10:07:42 -0600
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <203101c956f2$91781450$33014c0a@ws4>
References: <203101c956f2$91781450$33014c0a@ws4>
Message-ID: <CDF5931B-8C68-4D05-8D06-EA8095D8AC96@freeswitch.org>

So receive DTMF respond with more DTMF?

/b

On Dec 5, 2008, at 10:00 AM, Frank @ Impact wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.



From msc at freeswitch.org  Fri Dec  5 08:07:39 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 08:07:39 -0800
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
In-Reply-To: <1228488785.5481.4.camel@opos20.altron.lan>
References: <1228488785.5481.4.camel@opos20.altron.lan>
Message-ID: <579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>

Evgeniy,

I will need some time to digest all of this. I have an a104 but I  
don't have a solaris system for testing. I will report back as soon as  
I can.

-MC



On Dec 5, 2008, at 6:53 AM, Evgeniy Zolotov <zolotov at altron.ua> wrote:

>  Greetings!
>
> Question about possibility of the use FreeSWITCH for work with T1/E1
> streams under Sun Solaris 10 a bit clears up (Solaris 11 is in  
> condition
> of alpha-version and not suitable for the industrial use). But answers
> carry more negative sense.
>
> Start of T1/E1 under Sun Solaris has 2 stages: a) start of wanpipe's
> interface; b) make FreeSWITCH for Sun Solaris.
>
>
> a) For Linux Sangoma recommends
> (http://wiki.sangoma.com/wanpipe-freeswitch-install) installation of  
> new
> interface wanpipe-3.3.14.tgz, which has beta-status and placed at
> ftp://ftp.sangoma.com/linux/current_wanpipe.
>
> This installation and tests were successful.
>
> In this release (2007 y.) by Sangoma was added TDM API - native
> interface for FreeSwitch (YATE) ? which is absent in all previous
> releases of WANPIPE. In the Linux release 3.3.14 this variant was  
> named
> as ?TDM API?, in older releases - ?TDM Voice?.
>
> In this case FreeSWITCH works very good with E1/T1 cards
> A-101/102/104/108 without installation in zaptel system, and spans in
> configuration files of FreeSWITCH declared as [wanpipe#].
>
> But for Sun Solaris we found only
> drverftp://ftp.sangoma.com/Solaris/Beta/SVwanpipe-i386-5.10.pkgbeta,
> which dated 2007:
>
>> NAME=sangoma.com Wanpipe Driver
>
>> VERSION=1.1.0,REV=2007-07-16,
>
> Packet Svwanpipe-i386-5.10.pkg uses Svzaptel-i386-5.10.pkg, which  
> wasn't
> developed by Sangoma, but by little free
> communityhttp://www.solarisvoip.com/, source codes of this packet is
> herehttps://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk/
>
> The analysis of source codes shows that this project develops very
> slowly (there are no updates for about year), has very limited
> functionality and supports ( unlike original zaptel ) very limited  
> list
> of cards ( only one ;) - Digium Wildcard TE110P T1/PRI).
>
> Svwanpipe-i386-5.10.pkg supports only 64-bit Sun Solaris (on CD, which
> we get with Sangoma's cards, presents 32-bit driver and PDF document
> about installation under Sun Solaris ? but it dated 2001 ? 2002 yy 
> ).
>
> At first we have checked up installation with TDM Voice + zaptel (like
> for Asterisk) under Linux. We configured PRI spans as [zt] ... - such
> installation works good and we could do calls :
>
>
>> originate openzap/1/A/20000 &sleep(3)
>
>
>      * evidently that call retranslates from span 1 to span 2,
>        connected with cross-cable, and goes to extension 20000.
>
>
>
> b) About making FreeSwitch:
>
> Under Sun Solaris 10 with GCC makes FreeSWITCH core and most modules,
> except openzapand some others, because Sun Solaris 10 has GCC 3.4,
> but ./configure for openzap requires compatibility with ANSII 99.
>
> We have checked 3 different methods of makinf FreeSWITCH :
>
>
> 1. set up GCC 4.0.2 from CSW-repositaries (and all reguired for GCC
> *.pkg);
>
> 2. set up SunStudio 12 and do make cc/CC;
>
> 3.cross-compile FreeSwitch for Sun Solaris under Linux with CC- 
> options.
>
>
> In testing we used assembly SunStudio as 32-bit application (64-bit
> comes to the end with mistakes of assembly of some libraries, it is
> possible to correct for it easily, but we did not begin to specify  
> it).
>
>
> For testing cards A-104 we have repeated the same installation and
> configuring as under Linux (wanpipe TDM Voice + zaptel) on 4 different
> 64-bit servers under Sun Solaris 10:
>
>
>      * on 3 servers (2 of them manufactured by SUN) executed
>        successfully : # wanrouter start
>
>
>
> 4 wanrouter# interfaces was created; leds on spans, connected with
> cross-cable, becomes GREEN, i.e. synchronization T1/E1 presents (no
> alarms).
>
> On these servers FreeSWITCH correctly makes:
>
>
>> load mod_openzap
>
>
>      * but when we make :
>
>
>
>> originate openzap/1/A/20000 &sleep(3)
>
>
> - for 2 connected with crosss-cable spans (1 & 2) FreeSWITCH transmits
> PRI message (chan 1/31), but chan 2/31 doesn't receive this message
> (unlike under Linux) and call breaks after timeout.
>
> On 4-th server wanpipe doesn't even starts by mistake some IOCTL (i.e.
> at a command for device). Three ?working? servers is on AMD Opteron  
> Dual
> Core 2214(F), fourth is on ntel Xeon 3210.
>
> So, we supposes Svwanpipe-i386-5.10.pkg or Svzaptel-i386-5.10.pkg from
> Sangoma checked up a little and on some one processor and doesn't  
> heave
> up even on beta, as they are declared, at the best on alpha
>
>
> What's the reason of error???
>
>      * wanpipe doesn't work under Solaris?
>
>      * wrong working signalling with zaptel? (but same configuration
>        works good under Linux)
>
>      * wrong working of FreeSWITCH, which was built correctly, but
>        their work was violated?
>      *
>      *
> Thanks, Evgeniy.
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From frank at impactfax.com  Fri Dec  5 08:08:01 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 11:08:01 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <CDF5931B-8C68-4D05-8D06-EA8095D8AC96@freeswitch.org>
Message-ID: <205401c956f3$a60b4d50$33014c0a@ws4>

Yes.  listen in for 1 DTMF during a call and then signal back a
different DTMF.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
 
So receive DTMF respond with more DTMF?

/b

On Dec 5, 2008, at 10:00 AM, Frank @ Impact wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From Claudio.Cavalera at italtel.it  Fri Dec  5 08:09:12 2008
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Fri, 5 Dec 2008 17:09:12 +0100
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <203101c956f2$91781450$33014c0a@ws4>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0969622@BESONE.corp.dom>

freeswitch-users-bounces at lists.freeswitch.org wrote:
> After the tone is sent back out, we are done.  There is
> nothing left to
> do.


Maybe you can look at:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
Ciao,
Claudio


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From regs at kinetix.gr  Fri Dec  5 08:11:47 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 18:11:47 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
Message-ID: <493952C3.9060202@kinetix.gr>

Both legs are SIP. From non-registered endpoints (if of any use).

Michael S Collins wrote:
> I will do some research on this and let you know what I find out.  
> Question: are these internal calls or pstn or ?? Just curious about  
> your environment.
>
> Thanks,
> MC
>
>
>
> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>  
> wrote:
>
>   
>> The proto_specific_hangup_cause is missing on one of the two
>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>> When the callee hangs up it is missing from the b-leg CDR. Is this  
>> nornal?
>>
>> And another question : what piece of info could inform me about who
>> hanged up?
>>
>>
>> -- 
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From brian at freeswitch.org  Fri Dec  5 08:12:50 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 10:12:50 -0600
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
In-Reply-To: <579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>
References: <1228488785.5481.4.camel@opos20.altron.lan>
	<579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>
Message-ID: <43F6603A-909B-4B35-A04A-FE574814ECB9@freeswitch.org>

Does it list wanpipe TDM support on the Solaris builds of wanpipe?  I  
wasn't aware the TDM stuff was ported yet.

/b

On Dec 5, 2008, at 10:07 AM, Michael S Collins wrote:

> Evgeniy,
>
> I will need some time to digest all of this. I have an a104 but I
> don't have a solaris system for testing. I will report back as soon as
> I can.
>
> -MC



From regs at kinetix.gr  Fri Dec  5 08:17:07 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 18:17:07 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
Message-ID: <49395403.6080404@kinetix.gr>

I am sending this second email to request/suggest/enquire about 
something relevant :

Wouldn't it be useful to know which end of a specific call leg send the 
protocol
specific hangup cause? Otherwise it would be difficult to understand 
what really happened.



Michael S Collins wrote:
> I will do some research on this and let you know what I find out.  
> Question: are these internal calls or pstn or ?? Just curious about  
> your environment.
>
> Thanks,
> MC
>
>
>
> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>  
> wrote:
>
>   
>> The proto_specific_hangup_cause is missing on one of the two
>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>> When the callee hangs up it is missing from the b-leg CDR. Is this  
>> nornal?
>>
>> And another question : what piece of info could inform me about who
>> hanged up?
>>
>>
>> -- 
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From msc at freeswitch.org  Fri Dec  5 08:20:11 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 08:20:11 -0800
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <203101c956f2$91781450$33014c0a@ws4>
References: <203101c956f2$91781450$33014c0a@ws4>
Message-ID: <041F4629-11AD-4836-803F-9CD891454C3D@freeswitch.org>

Will the call be terminated at that point or does it need to continue?  
I do know that the tone_detect app can listen for a dtmf from either  
direction and can trigger execution of another app/extension/etc.  
However, I've never tried it on a bridged call, so I'm curious to see  
what would happen. The other question I would need to research is what  
would happen if the dtmf was sent rfc2833 style.

Hop on the wiki and look at tone_detect while I research the other  
questions and we will see what we can come up with.

-MC


On Dec 5, 2008, at 8:00 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-
>
> What would need to happen after the tone is sent back out? Also, would
> this be part of something like an IVR?
>
> -MC
>
>
> On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>
> wrote:
>
>>
>> Is there any dialplan instructions that could be added that would sit
>> and listen during a call for a tone (a key press, say 2) and when FS
>> hears that tone, then FS can broadcast another key tone (say 6) back
>> to
>> the channels?
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mike at jerris.com  Fri Dec  5 08:22:53 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 11:22:53 -0500
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
In-Reply-To: <43F6603A-909B-4B35-A04A-FE574814ECB9@freeswitch.org>
References: <1228488785.5481.4.camel@opos20.altron.lan>
	<579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>
	<43F6603A-909B-4B35-A04A-FE574814ECB9@freeswitch.org>
Message-ID: <21906AF3-7B10-4097-88AC-48F245E0498A@jerris.com>

Last I spoke to doug at sangoma, solaris support is still not in their  
platform abstraction lib (there are drivers).  Please contact sangoma  
sales and request this.

Mike.

p.s. make sure to tell them its for FreeSWITCH


On Dec 5, 2008, at 11:12 AM, Brian West wrote:

> Does it list wanpipe TDM support on the Solaris builds of wanpipe?  I
> wasn't aware the TDM stuff was ported yet.
>
> /b
>
> On Dec 5, 2008, at 10:07 AM, Michael S Collins wrote:
>
>> Evgeniy,
>>
>> I will need some time to digest all of this. I have an a104 but I
>> don't have a solaris system for testing. I will report back as soon  
>> as
>> I can.
>>
>> -MC
>


From brian at freeswitch.org  Fri Dec  5 08:23:13 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 10:23:13 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493952C3.9060202@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<493952C3.9060202@kinetix.gr>
Message-ID: <1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>

Did you say what SVN rev you're running.

/b

On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:

> Both legs are SIP. From non-registered endpoints (if of any use).



From frank at impactfax.com  Fri Dec  5 08:29:57 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 11:29:57 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <041F4629-11AD-4836-803F-9CD891454C3D@freeswitch.org>
Message-ID: <207d01c956f6$b6165430$33014c0a@ws4>

The call should continue after FS hears the key press and responds with
its own key press tone.   Then the call just continues on.  They key
press from one of the parties would come some time after the call is
bridged.  Maybe some 10 or 20 seconds into the call for example.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins
 
Will the call be terminated at that point or does it need to continue?  
I do know that the tone_detect app can listen for a dtmf from either  
direction and can trigger execution of another app/extension/etc.  
However, I've never tried it on a bridged call, so I'm curious to see  
what would happen. The other question I would need to research is what  
would happen if the dtmf was sent rfc2833 style.

Hop on the wiki and look at tone_detect while I research the other  
questions and we will see what we can come up with.

-MC


On Dec 5, 2008, at 8:00 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-
>
> What would need to happen after the tone is sent back out? Also, would
> this be part of something like an IVR?
>
> -MC
>
>
> On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>
> wrote:
>
>>
>> Is there any dialplan instructions that could be added that would sit
>> and listen during a call for a tone (a key press, say 2) and when FS
>> hears that tone, then FS can broadcast another key tone (say 6) back
>> to
>> the channels?
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From frank at impactfax.com  Fri Dec  5 08:32:54 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 11:32:54 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <041F4629-11AD-4836-803F-9CD891454C3D@freeswitch.org>
Message-ID: <208801c956f7$1f80dee0$33014c0a@ws4>

Looks like tone detect might do it.  But..

If so, What frequency would we use for particular keys?  

Will tone_Detect sniff both legs or would we just do both r and w on the
called leg?

Can the tone_Detect timeout just be a very large number or can we leave
out the timeout value so there is no timeout?

Could the trigger from tone Detect do a gentone for a certain key?

Not much on the wiki on the mod.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins


Hop on the wiki and look at tone_detect while I research the other  
questions and we will see what we can come up with.

-MC


On Dec 5, 2008, at 8:00 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-
>
> What would need to happen after the tone is sent back out? Also, would
> this be part of something like an IVR?
>
> -MC
>
>
> On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>
> wrote:
>
>>
>> Is there any dialplan instructions that could be added that would sit
>> and listen during a call for a tone (a key press, say 2) and when FS
>> hears that tone, then FS can broadcast another key tone (say 6) back
>> to
>> the channels?
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_______________________________________________
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From cstomi.levlist at gmail.com  Fri Dec  5 08:34:45 2008
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Fri, 05 Dec 2008 17:34:45 +0100
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
References: <20677406.post@talk.nabble.com>	<BA08ED43-200A-4037-935D-B15716DC09AD@jerris.com>	<191c3a030811250600n5ba54fc0qb219b09e19726adf@mail.gmail.com>	<20687620.post@talk.nabble.com>	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>	<20689832.post@talk.nabble.com>	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
Message-ID: <49395825.2010008@gmail.com>

Hello,

I have the same problem,

I don't understand the difference between

progress_timeout
originate_timeout
call_timeout.

I log timelimit_sec in switch_ivr_originate function and it seems,
if I set call_timeout then, timelimit_sec will be this value
if I set originate_timeout then timelimit_sec will be this value. maybe 
this is for backward compat?

originate_timeout as in the wiki:
 "Determines how long FreeSwitch is going to wait for a response from 
the invite message sent to the gateway. "

I quess this would be an 100 reply.

But how could I set a timeout for 200? I mean timeout for an answer.
which variable would control this?
I quess it was call_timeout previosly.
Please explain me this timeout variables

Thanks,
Tamas

Michael Collins ?rta:
> FYI, it is on the channel variables page but it's in a crazy place under
> "unknown functionality" which is silly.
> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
>
> Anyway, I've got wiki cleaning on my to-do list and I'll start in earnest
> next month when I have some time...
>
> -MC
>
> On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>   
>>> I used "call-timeout" because that's how I understood it from the Wiki.
>>>  (?)
>>>
>>>       
>> Yep, that's all that there is on the wiki. Unfortunately the channel
>> variables page is one of many in need of some attention. I will add
>> "originate_timeout" right away. The only question remaining is what, if
>> anything, does call_timeout do? That channel variable is in the source code
>> but I don't know exactly what it does.
>>
>> -MC
>>
>>
>>     
>>> --
>>> View this message in context:
>>> http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>     
>
>   
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From msc at freeswitch.org  Fri Dec  5 08:35:40 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 08:35:40 -0800
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <49395403.6080404@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<49395403.6080404@kinetix.gr>
Message-ID: <5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>

Makes sense. I will look into this.
-MC


On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>  
wrote:

> I am sending this second email to request/suggest/enquire about  
> something relevant :
>
> Wouldn't it be useful to know which end of a specific call leg send  
> the protocol
> specific hangup cause? Otherwise it would be difficult to understand  
> what really happened.
>
>
>
> Michael S Collins wrote:
>>
>> I will do some research on this and let you know what I find out.
>> Question: are these internal calls or pstn or ?? Just curious about
>> your environment.
>>
>> Thanks,
>> MC
>>
>>
>>
>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>> wrote:
>>
>>
>>> The proto_specific_hangup_cause is missing on one of the two
>>> call legs. When the caller hangs up it is missing from the a-leg  
>>> CDR.
>>> When the callee hangs up it is missing from the b-leg CDR. Is this
>>> nornal?
>>>
>>> And another question : what piece of info could inform me about who
>>> hanged up?
>>>
>>>
>>> -- 
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> -- 
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Fri Dec  5 08:37:26 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 10:37:26 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<493952C3.9060202@kinetix.gr>
	<1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
Message-ID: <191c3a030812050837n4374e96fya71588a028869dc5@mail.gmail.com>

It's easy enough to set the value on both legs try r10614
It was only set on the opposing leg before but since it's harmless to set it
on both i did it for you.


On Fri, Dec 5, 2008 at 10:23 AM, Brian West <brian at freeswitch.org> wrote:

> Did you say what SVN rev you're running.
>
> /b
>
> On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
>
> > Both legs are SIP. From non-registered endpoints (if of any use).
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Fri Dec  5 08:41:16 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 10:41:16 -0600
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <49395825.2010008@gmail.com>
References: <20677406.post@talk.nabble.com>
	<BA08ED43-200A-4037-935D-B15716DC09AD@jerris.com>
	<191c3a030811250600n5ba54fc0qb219b09e19726adf@mail.gmail.com>
	<20687620.post@talk.nabble.com>
	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>
	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>
	<20689832.post@talk.nabble.com>
	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
	<49395825.2010008@gmail.com>
Message-ID: <191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>

call_timeout is only used if you are bridging 2 channels where one or both
of them is still unanswered.

what you want to use is originate_timeout and forget about call_timeout

you also have
leg_timeout and leg_progress_timeout both to be set in the {}
that do the timeout from the perspective of the new channel leg instead of
the caller leg.



On Fri, Dec 5, 2008 at 10:34 AM, Tamas Cseke <cstomi.levlist at gmail.com>wrote:

> Hello,
>
> I have the same problem,
>
> I don't understand the difference between
>
> progress_timeout
> originate_timeout
> call_timeout.
>
> I log timelimit_sec in switch_ivr_originate function and it seems,
> if I set call_timeout then, timelimit_sec will be this value
> if I set originate_timeout then timelimit_sec will be this value. maybe
> this is for backward compat?
>
> originate_timeout as in the wiki:
>  "Determines how long FreeSwitch is going to wait for a response from
> the invite message sent to the gateway. "
>
> I quess this would be an 100 reply.
>
> But how could I set a timeout for 200? I mean timeout for an answer.
> which variable would control this?
> I quess it was call_timeout previosly.
> Please explain me this timeout variables
>
> Thanks,
> Tamas
>
> Michael Collins ?rta:
> > FYI, it is on the channel variables page but it's in a crazy place under
> > "unknown functionality" which is silly.
> > http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
> >
> > Anyway, I've got wiki cleaning on my to-do list and I'll start in earnest
> > next month when I have some time...
> >
> > -MC
> >
> > On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> >>> I used "call-timeout" because that's how I understood it from the Wiki.
> >>>  (?)
> >>>
> >>>
> >> Yep, that's all that there is on the wiki. Unfortunately the channel
> >> variables page is one of many in need of some attention. I will add
> >> "originate_timeout" right away. The only question remaining is what, if
> >> anything, does call_timeout do? That channel variable is in the source
> code
> >> but I don't know exactly what it does.
> >>
> >> -MC
> >>
> >>
> >>
> >>> --
> >>> View this message in context:
> >>>
> http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.html
> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>>
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From regs at kinetix.gr  Fri Dec  5 08:42:50 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 18:42:50 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<493952C3.9060202@kinetix.gr>
	<1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
Message-ID: <49395A0A.7070103@kinetix.gr>

FreeSWITCH Version 1.0.trunk (10579)

Brian West wrote:
> Did you say what SVN rev you're running.
>
> /b
>
> On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
>
>   
>> Both legs are SIP. From non-registered endpoints (if of any use).
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From jbr at consiglia.dk  Fri Dec  5 08:43:44 2008
From: jbr at consiglia.dk (Jon Bruel)
Date: Fri, 5 Dec 2008 17:43:44 +0100
Subject: [Freeswitch-users] How to force FS using the hosts entry in
	outgoing gateway calls
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>

For the configuration of a gateway I need to use a specific proxy domain name before the server (Covergence SBC with a BroadWorks Application Server behind) accepts calls. The twist is that the right proxy name points the wrong IP-address (the voicemail server for the account). I have tried to overrule this by adding a host entry (Linux). When I ping to the domain name I get the right address (the one from the host table), but the FS uses the address from the DNS lookup, not the address from the host table. What can I do to force the FS using the entry from the host table? Thanks /Jon.

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From anthony.minessale at gmail.com  Fri Dec  5 08:53:53 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 10:53:53 -0600
Subject: [Freeswitch-users] How to force FS using the hosts entry in
	outgoing gateway calls
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>
Message-ID: <191c3a030812050853p5257eeefvb421ff045ad3ef9d@mail.gmail.com>

set proxy to be the correct hostname and set register-proxy param to be the
correct IP


On Fri, Dec 5, 2008 at 10:43 AM, Jon Bruel <jbr at consiglia.dk> wrote:

>  For the configuration of a gateway I need to use a specific proxy domain
> name before the server (Covergence SBC with a BroadWorks Application Server
> behind) accepts calls. The twist is that the right proxy name points the
> wrong IP-address (the voicemail server for the account). I have tried to
> overrule this by adding a host entry (Linux). When I ping to the domain name
> I get the right address (the one from the host table), but the FS uses the
> address from the DNS lookup, not the address from the host table. What can I
> do to force the FS using the entry from the host table? Thanks /Jon.
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From Prometheus001 at gmx.net  Fri Dec  5 09:08:46 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Dec 2008 18:08:46 +0100
Subject: [Freeswitch-users] Event_Socket: How to determine
 the	ogininater uuid before an outgoing call is set up?
In-Reply-To: <F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>
References: <4939168E.9020400@gmx.net>
	<F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>
Message-ID: <4939601E.7070601@gmx.net>

I am a step further, When I set the cid-name then I can access the data
dring
channel_outgoing
channel_originate
channel_progress
channel_answer

However setting the caller_caller_id_number might be better.

This is the originate request:
<?xml version="1.0" ?>
<methodCall>
<methodName>freeswitch.api</methodName>
<params>
<param><value><string>bgapi</string></value></param>
<param><value><string>originate
{other_unique_id=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_name=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_number=000,ignore_early_media=true}user/1001 at siplocal.safecomm.ch
&amp;transfer(5002)</string></value></param>
</params>
</methodCall>

Answer:
<?xml version="1.0" encoding="UTF-8"?>.
<methodResponse>.
<params>.
<param><value><string>+OK Job-UUID: 0856d3ec-c2e9-11dd-85f2-75efbd1bca02
</string></value></param>.
</params>.
</methodResponse>.

By the way: The Job-UUID is different from the channel uuid, so it
cannot be used for my issue.

Best regards
Peter

Michael S Collins schrieb:
> What is your originate string?
> -MC
>
>
> On Dec 5, 2008, at 3:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>
>   
>> I am building an IVR application where an incoming call is  
>> initiating an
>> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
>> incoming channel) at originate time, I am able to reference to the
>> incomig call, once the outgoing call is set up. So the outgoing call  
>> can
>> see the uuid of the incoming call who has originated the outgoing  
>> call.
>> This is needed for bridging the 2 calls together.
>>
>> However I want to control also the call setup process (see, if the
>> outgoing call is ringing etc.). At call setup time, when I parse the
>> channel_originate ,channel_outgoing and channel_progress events, I
>> cannot see any reference to the incoming call (variable_other_uuid is
>> not set). I suspect that variables are only passed once the outgoing
>> channel is set up.
>>
>> Has anybody an idea, how I may get the uuid of the originating uuid in
>> the outgoing call at call setup?
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From msc at freeswitch.org  Fri Dec  5 09:38:32 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 09:38:32 -0800
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <208801c956f7$1f80dee0$33014c0a@ws4>
References: <041F4629-11AD-4836-803F-9CD891454C3D@freeswitch.org>
	<208801c956f7$1f80dee0$33014c0a@ws4>
Message-ID: <87f2f3b90812050938s120f801dy29a95d02f601f89a@mail.gmail.com>

On Fri, Dec 5, 2008 at 8:32 AM, Frank @ Impact <frank at impactfax.com> wrote:
> Looks like tone detect might do it.  But..
>
> If so, What frequency would we use for particular keys?
>

http://en.wikipedia.org/wiki/DTMF#Keypad


> Will tone_Detect sniff both legs or would we just do both r and w on the
> called leg?
>

Just do both r and w.

> Can the tone_Detect timeout just be a very large number or can we leave
> out the timeout value so there is no timeout?

I know you can set it to a large number; I've never tried a "forever"
tone_detect. I'll check it out.

>
> Could the trigger from tone Detect do a gentone for a certain key?
>

I don't believe so. This is where the bind_meta_app functionality is
more applicable. The dialplan isn't really the best place to handle
"events" like this. (Event socket would be better if you can swing
that, but I think maybe a workaround is doable with just the dialplan
and some creativity.)

> Not much on the wiki on the mod.

My bad. I'm working on it. :) In the meantime grab those dtmf
frequency values and set up a test extension in your dialplan. Put a
tone_detect app in that test ext. (You could have the ext set tone
detect, sleep 10 seconds, do info app then hangup.) Then call the test
extension, press a few keys, then wait for the info app to dump. If
you have the tone_detect set a chan variable when a dtmf is pressed
then you'll see it in your info app dump. BTW, MikeJ reminded me about
the start_dtmf/stop_dtmf apps:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf

Those might be necessary if your dtmf's are not already in-band.

Here's a sample extension you could try for testing, dialing 9990:

<extension name="dtmf_detect_test">
    <condition field="destination_number" expression="^9990$">
        <action application="answer"/>
        <action application="tone_detect" data="DTMF1 697,1209 rw
+10000 set DTMF1=true"/>  <!-- detect DTMF 1 key-->
        <action application="sleep" data="10000"/>
        <action application="info"/>
        <action application="hangup"/>
    </condition>
</extension>

Give that a try and at least see if you can detect the tones...
-MC


From msc at freeswitch.org  Fri Dec  5 09:50:37 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 09:50:37 -0800
Subject: [Freeswitch-users] Event_Socket: How to determine the
	ogininater uuid before an outgoing call is set up?
In-Reply-To: <4939601E.7070601@gmx.net>
References: <4939168E.9020400@gmx.net>
	<F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>
	<4939601E.7070601@gmx.net>
Message-ID: <87f2f3b90812050950r2cfe7966h6be1146aa828b5ad@mail.gmail.com>

Peter,

thanks, I will ruminate on this and get back with you as soon as I can.
-MC

On Fri, Dec 5, 2008 at 9:08 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I am a step further, When I set the cid-name then I can access the data
> dring
> channel_outgoing
> channel_originate
> channel_progress
> channel_answer
>
> However setting the caller_caller_id_number might be better.
>
> This is the originate request:
> <?xml version="1.0" ?>
> <methodCall>
> <methodName>freeswitch.api</methodName>
> <params>
> <param><value><string>bgapi</string></value></param>
> <param><value><string>originate
> {other_unique_id=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_name=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_number=000,ignore_early_media=true}user/1001 at siplocal.safecomm.ch
> &amp;transfer(5002)</string></value></param>
> </params>
> </methodCall>
>
> Answer:
> <?xml version="1.0" encoding="UTF-8"?>.
> <methodResponse>.
> <params>.
> <param><value><string>+OK Job-UUID: 0856d3ec-c2e9-11dd-85f2-75efbd1bca02
> </string></value></param>.
> </params>.
> </methodResponse>.
>
> By the way: The Job-UUID is different from the channel uuid, so it
> cannot be used for my issue.
>
> Best regards
> Peter
>
> Michael S Collins schrieb:
>> What is your originate string?
>> -MC
>>
>>
>> On Dec 5, 2008, at 3:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>
>>> I am building an IVR application where an incoming call is
>>> initiating an
>>> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
>>> incoming channel) at originate time, I am able to reference to the
>>> incomig call, once the outgoing call is set up. So the outgoing call
>>> can
>>> see the uuid of the incoming call who has originated the outgoing
>>> call.
>>> This is needed for bridging the 2 calls together.
>>>
>>> However I want to control also the call setup process (see, if the
>>> outgoing call is ringing etc.). At call setup time, when I parse the
>>> channel_originate ,channel_outgoing and channel_progress events, I
>>> cannot see any reference to the incoming call (variable_other_uuid is
>>> not set). I suspect that variables are only passed once the outgoing
>>> channel is set up.
>>>
>>> Has anybody an idea, how I may get the uuid of the originating uuid in
>>> the outgoing call at call setup?
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Fri Dec  5 09:57:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 11:57:51 -0600
Subject: [Freeswitch-users] Event_Socket: How to determine the
	ogininater uuid before an outgoing call is set up?
In-Reply-To: <87f2f3b90812050950r2cfe7966h6be1146aa828b5ad@mail.gmail.com>
References: <4939168E.9020400@gmx.net>
	<F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>
	<4939601E.7070601@gmx.net>
	<87f2f3b90812050950r2cfe7966h6be1146aa828b5ad@mail.gmail.com>
Message-ID: <191c3a030812050957v43ccaf33r5ba100cdcbf4e5e7@mail.gmail.com>

job-uuid can be used to match the BACKGROUND_JOB event which will
have the output of the originate command in the body.

since you are using bgapi it goes asyncronous and must deliver the reply to
you via the event interface.


On Fri, Dec 5, 2008 at 11:50 AM, Michael Collins <msc at freeswitch.org> wrote:

> Peter,
>
> thanks, I will ruminate on this and get back with you as soon as I can.
> -MC
>
> On Fri, Dec 5, 2008 at 9:08 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> > I am a step further, When I set the cid-name then I can access the data
> > dring
> > channel_outgoing
> > channel_originate
> > channel_progress
> > channel_answer
> >
> > However setting the caller_caller_id_number might be better.
> >
> > This is the originate request:
> > <?xml version="1.0" ?>
> > <methodCall>
> > <methodName>freeswitch.api</methodName>
> > <params>
> > <param><value><string>bgapi</string></value></param>
> > <param><value><string>originate
> >
> {other_unique_id=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_name=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_number=000,ignore_early_media=true}user/
> 1001 at siplocal.safecomm.ch
> > &amp;transfer(5002)</string></value></param>
> > </params>
> > </methodCall>
> >
> > Answer:
> > <?xml version="1.0" encoding="UTF-8"?>.
> > <methodResponse>.
> > <params>.
> > <param><value><string>+OK Job-UUID: 0856d3ec-c2e9-11dd-85f2-75efbd1bca02
> > </string></value></param>.
> > </params>.
> > </methodResponse>.
> >
> > By the way: The Job-UUID is different from the channel uuid, so it
> > cannot be used for my issue.
> >
> > Best regards
> > Peter
> >
> > Michael S Collins schrieb:
> >> What is your originate string?
> >> -MC
> >>
> >>
> >> On Dec 5, 2008, at 3:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> >>
> >>
> >>> I am building an IVR application where an incoming call is
> >>> initiating an
> >>> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
> >>> incoming channel) at originate time, I am able to reference to the
> >>> incomig call, once the outgoing call is set up. So the outgoing call
> >>> can
> >>> see the uuid of the incoming call who has originated the outgoing
> >>> call.
> >>> This is needed for bridging the 2 calls together.
> >>>
> >>> However I want to control also the call setup process (see, if the
> >>> outgoing call is ringing etc.). At call setup time, when I parse the
> >>> channel_originate ,channel_outgoing and channel_progress events, I
> >>> cannot see any reference to the incoming call (variable_other_uuid is
> >>> not set). I suspect that variables are only passed once the outgoing
> >>> channel is set up.
> >>>
> >>> Has anybody an idea, how I may get the uuid of the originating uuid in
> >>> the outgoing call at call setup?
> >>>
> >>> Best regards
> >>> Peter
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mehdi.chaabouni at gmail.com  Fri Dec  5 10:23:24 2008
From: mehdi.chaabouni at gmail.com (mehdix)
Date: Fri, 5 Dec 2008 10:23:24 -0800 (PST)
Subject: [Freeswitch-users]  Provider: Junction Networks
Message-ID: <20859688.post@talk.nabble.com>


I've got a problem with configuring a SIP trunk from Junction Networks with
FS: it only works for a few minutes then the line is dropped.

I tried Unlimitel with no problem.

Any Ideas?

Thanks
-- 
View this message in context: http://www.nabble.com/Provider%3A-Junction-Networks-tp20859688p20859688.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Fri Dec  5 10:23:37 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 10:23:37 -0800
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>
References: <20677406.post@talk.nabble.com>
	<191c3a030811250600n5ba54fc0qb219b09e19726adf@mail.gmail.com>
	<20687620.post@talk.nabble.com>
	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>
	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>
	<20689832.post@talk.nabble.com>
	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
	<49395825.2010008@gmail.com>
	<191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>
Message-ID: <87f2f3b90812051023v3aebd168r2d92ae44531d93bb@mail.gmail.com>

On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> call_timeout is only used if you are bridging 2 channels where one or both
> of them is still unanswered.
>
> what you want to use is originate_timeout and forget about call_timeout
>
> you also have
> leg_timeout and leg_progress_timeout both to be set in the {}
> that do the timeout from the perspective of the new channel leg instead of
> the caller leg.
>

I will make sure that the wiki reflects these explanations properly.
-MC


From brian at freeswitch.org  Fri Dec  5 10:30:46 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 10:30:46 -0800
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <20859688.post@talk.nabble.com>
References: <20859688.post@talk.nabble.com>
Message-ID: <04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>

What is the hangup cause?

/b

On Dec 5, 2008, at 10:23 AM, mehdix wrote:

> Any Ideas?



From jan.kubr at gmail.com  Fri Dec  5 10:33:21 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Fri, 5 Dec 2008 19:33:21 +0100
Subject: [Freeswitch-users] DTMF from cell phones
Message-ID: <698401620812051033lf758838m733191df67143cea@mail.gmail.com>

> > no solution. I have a similar problem, when calling Freeswitch from my
> > cell phone (via a SIP provider), sometimes DTMF is not recognized
>> The important thing to note is that when using
>> a SIP softphone (X-Lite) I have never had this problem, DTMF is


> So i guess that using latest version with a few changes in your config
> should work unless there's any other issue related to your sip provider
> ( PSTN / Media Gateway ), on this case you can get some captures of
> sip/rtp traffic to check SDP and rtp Marks.

I tried trunk and the values for the variables (all except
rtp-timer-name=none are already default in trunk), but only two things
are different:
1. When I press a key, the read app seem to always terminate, but not
always the dtmf is captured in a variable.
2. The read app seems to ignore the variable name parameter: calling
it with "1 1 104.wav choice_181152 10000 #" doesn't put the digit in
variable_choice_181152, but to dmtf_digit. Why is that?

> If it is coming from the sip provider as rfc 2833 dtmf, they are
> probably doing inband detection and failing at it.  If you look at an
> rtp dump you can confirm this.  If this is the case, there is nothing
> you can do on the FreeSWITCH side and the provider will have to fix it.

But the call goes through the same SIP provider even when using the
soft phone and there it works fine. The difference might be that then
it is SIP to SIP within the same provider.. How do I do the RTP dump?

Also I should have mentioned that DTMF is not captured only DURING the
file is being played. It is always captured correctly when I wait
until the playback is finished. Does this sound familiar? I thought
this would be somet obvious misconfiguration on my side.


Jan


From zolotov at altron.ua  Fri Dec  5 10:36:02 2008
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Fri, 05 Dec 2008 20:36:02 +0200
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
In-Reply-To: <21906AF3-7B10-4097-88AC-48F245E0498A@jerris.com>
References: <1228488785.5481.4.camel@opos20.altron.lan>
	<579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>
	<43F6603A-909B-4B35-A04A-FE574814ECB9@freeswitch.org>
	<21906AF3-7B10-4097-88AC-48F245E0498A@jerris.com>
Message-ID: <1228502162.5481.29.camel@opos20.altron.lan>

  Thanks to all for their answers.

1. to Michael Collins

> >> I will need some time to digest all of this. I have an a104 but I
> >> don't have a solaris system for testing. I will report back as soon  
> >> as I can

 We with impatience will wait for results of your tests. If there will
be any questions - we with pleasure will help you.

2. to Brian West

> > Does it list wanpipe TDM support on the Solaris builds of wanpipe?  I
> > wasn't aware the TDM stuff was ported yet.

There are 2 kind of TDM support into Sangoma. Into Linux installation
menu (ncurses) they named :
  #2 "TDM Voice", where signalling carries out with zaptel and
  #8 "TDM API"(libsangoma), where signalling carries out without zaptel,
this is native interface, which is used by FreeSWITCH and Yate.

In Svwanpipe-i386-5.10.pkg (for Sun Solaris) present "TDM Voice", but 
absent "TDM API". Despite the fact that "TDM Voice" is present, seems
it works incorrectly (it works good under Linux, but not Sun Solaris).

3. to Michael Jerris

> Last I spoke to doug at sangoma, solaris support is still not in their  
> platform abstraction lib (there are drivers).  Please contact sangoma  
> sales and request this.
> 
> Mike.
> 
> p.s. make sure to tell them its for FreeSWITCH

Michael, I sent the same messages to Sangona, but they similar ignore
them, because I have not received any answer from them.




? ???, 05/12/2008 ? 11:22 -0500, Michael Jerris ?????:
> Last I spoke to doug at sangoma, solaris support is still not in their  
> platform abstraction lib (there are drivers).  Please contact sangoma  
> sales and request this.
> 
> Mike.
> 
> p.s. make sure to tell them its for FreeSWITCH
> 
> 
> On Dec 5, 2008, at 11:12 AM, Brian West wrote:
> 
> > Does it list wanpipe TDM support on the Solaris builds of wanpipe?  I
> > wasn't aware the TDM stuff was ported yet.
> >
> > /b
> >
> > On Dec 5, 2008, at 10:07 AM, Michael S Collins wrote:
> >
> >> Evgeniy,
> >>
> >> I will need some time to digest all of this. I have an a104 but I
> >> don't have a solaris system for testing. I will report back as soon  
> >> as
> >> I can.
> >>
> >> -MC
> >
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gservat at gmail.com  Fri Dec  5 10:38:14 2008
From: gservat at gmail.com (Gonzalo Servat)
Date: Fri, 5 Dec 2008 16:38:14 -0200
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <87f2f3b90812051023v3aebd168r2d92ae44531d93bb@mail.gmail.com>
References: <20677406.post@talk.nabble.com> <20687620.post@talk.nabble.com>
	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>
	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>
	<20689832.post@talk.nabble.com>
	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
	<49395825.2010008@gmail.com>
	<191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>
	<87f2f3b90812051023v3aebd168r2d92ae44531d93bb@mail.gmail.com>
Message-ID: <dcc007e10812051038j7d2ef8eaxde73462c2ba4b1c4@mail.gmail.com>

On Fri, Dec 5, 2008 at 4:23 PM, Michael Collins <msc at freeswitch.org> wrote:

> On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > call_timeout is only used if you are bridging 2 channels where one or
> both
> > of them is still unanswered.
> >
> > what you want to use is originate_timeout and forget about call_timeout
> >
> > you also have
> > leg_timeout and leg_progress_timeout both to be set in the {}
> > that do the timeout from the perspective of the new channel leg instead
> of
> > the caller leg.
> >
>
> I will make sure that the wiki reflects these explanations properly.
>

Excellent :) I'm still wondering not 100% clear on the exact difference(s)
between call_timeout and originate_timeout ...

- Gonzalo
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From regs at kinetix.gr  Fri Dec  5 10:43:52 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 20:43:52 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <191c3a030812050837n4374e96fya71588a028869dc5@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<493952C3.9060202@kinetix.gr>	<1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
	<191c3a030812050837n4374e96fya71588a028869dc5@mail.gmail.com>
Message-ID: <49397668.80808@kinetix.gr>

I tested it and it works fine but it got me thinking...

Is just a copy of the cause to the other leg the correct way
to do it? Couldn't the two call legs hang up with different causes?
Especially when I could override the cause before it got send
to the e.g. calling side using e.g. the hangup command?

To make myself clear : I could have the b-leg (in a bridge hangup)
sending me a user busy code and I could send a circuit/channel
unavailable to my caller (a-leg), let's
say because I don't trust my terminator (b-leg) and his codes and I want to
enforce another one and send it to my originator so that he could retry 
another
carrier.

What do you think?


Anthony Minessale wrote:
> It's easy enough to set the value on both legs try r10614
> It was only set on the opposing leg before but since it's harmless to 
> set it on both i did it for you.
>
>
> On Fri, Dec 5, 2008 at 10:23 AM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     Did you say what SVN rev you're running.
>
>     /b
>
>     On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
>
>     > Both legs are SIP. From non-registered endpoints (if of any use).
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From anthony.minessale at gmail.com  Fri Dec  5 10:46:05 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 12:46:05 -0600
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <dcc007e10812051038j7d2ef8eaxde73462c2ba4b1c4@mail.gmail.com>
References: <20677406.post@talk.nabble.com>
	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>
	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>
	<20689832.post@talk.nabble.com>
	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
	<49395825.2010008@gmail.com>
	<191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>
	<87f2f3b90812051023v3aebd168r2d92ae44531d93bb@mail.gmail.com>
	<dcc007e10812051038j7d2ef8eaxde73462c2ba4b1c4@mail.gmail.com>
Message-ID: <191c3a030812051046t5557da06w67d2cf3c2f09d657@mail.gmail.com>

forget call_timout it's your best bet

it's depricated.


On Fri, Dec 5, 2008 at 12:38 PM, Gonzalo Servat <gservat at gmail.com> wrote:

> On Fri, Dec 5, 2008 at 4:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > call_timeout is only used if you are bridging 2 channels where one or
>> both
>> > of them is still unanswered.
>> >
>> > what you want to use is originate_timeout and forget about call_timeout
>> >
>> > you also have
>> > leg_timeout and leg_progress_timeout both to be set in the {}
>> > that do the timeout from the perspective of the new channel leg instead
>> of
>> > the caller leg.
>> >
>>
>> I will make sure that the wiki reflects these explanations properly.
>>
>
> Excellent :) I'm still wondering not 100% clear on the exact difference(s)
> between call_timeout and originate_timeout ...
>
> - Gonzalo
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Dec  5 10:53:00 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 12:53:00 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <49397668.80808@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<493952C3.9060202@kinetix.gr>
	<1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
	<191c3a030812050837n4374e96fya71588a028869dc5@mail.gmail.com>
	<49397668.80808@kinetix.gr>
Message-ID: <191c3a030812051053g266828c8w4de90e8abdd816e0@mail.gmail.com>

This variable is to specifically document the protocol specific last status
cause.
So you can know what the status was when you got a BYE or final response to
invite in the case of sip.
That's all it's for.





On Fri, Dec 5, 2008 at 12:43 PM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  I tested it and it works fine but it got me thinking...
>
> Is just a copy of the cause to the other leg the correct way
> to do it? Couldn't the two call legs hang up with different causes?
> Especially when I could override the cause before it got send
> to the e.g. calling side using e.g. the hangup command?
>
> To make myself clear : I could have the b-leg (in a bridge hangup)
> sending me a user busy code and I could send a circuit/channel
> unavailable to my caller (a-leg), let's
> say because I don't trust my terminator (b-leg) and his codes and I want to
> enforce another one and send it to my originator so that he could retry
> another
> carrier.
>
> What do you think?
>
>
> Anthony Minessale wrote:
>
> It's easy enough to set the value on both legs try r10614
> It was only set on the opposing leg before but since it's harmless to set
> it on both i did it for you.
>
>
> On Fri, Dec 5, 2008 at 10:23 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Did you say what SVN rev you're running.
>>
>> /b
>>
>> On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
>>
>> > Both legs are SIP. From non-registered endpoints (if of any use).
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mehdi.chaabouni at gmail.com  Fri Dec  5 10:59:35 2008
From: mehdi.chaabouni at gmail.com (MEHDi CHAABOUNi)
Date: Fri, 5 Dec 2008 13:59:35 -0500
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
Message-ID: <c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>

Actually, i did not mean that the line is dropped during a call...
FS is configured to accept calls from the Junction Networks SIP trunk to
make an audio conference.
When I start FS and I dial the number all is working fine. But, if I wait
for a couple of minutes and then make my call I get an error recorded
message saying that the number is not in service...




On Fri, Dec 5, 2008 at 1:30 PM, Brian West <brian at freeswitch.org> wrote:

> What is the hangup cause?
>
> /b
>
> On Dec 5, 2008, at 10:23 AM, mehdix wrote:
>
> > Any Ideas?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Dec  5 11:05:14 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 11:05:14 -0800
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
Message-ID: <6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>

But you don't see the invite hitting FreeSWITCH?  And you're behind  
NAT?  Make it register every 30 seconds instead of the default 3600

/b

On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:

> Actually, i did not mean that the line is dropped during a call...
> FS is configured to accept calls from the Junction Networks SIP  
> trunk to make an audio conference.
> When I start FS and I dial the number all is working fine. But, if I  
> wait for a couple of minutes and then make my call I get an error  
> recorded message saying that the number is not in service...



From msc at freeswitch.org  Fri Dec  5 11:08:21 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 11:08:21 -0800
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
Message-ID: <87f2f3b90812051108v1ebb8809u25364323d9dfa6ad@mail.gmail.com>

Can you hit F8 and capture the debug output when making a call?
That'll help us see what's going on.
-MC

On Fri, Dec 5, 2008 at 10:59 AM, MEHDi CHAABOUNi
<mehdi.chaabouni at gmail.com> wrote:
> Actually, i did not mean that the line is dropped during a call...
> FS is configured to accept calls from the Junction Networks SIP trunk to
> make an audio conference.
> When I start FS and I dial the number all is working fine. But, if I wait
> for a couple of minutes and then make my call I get an error recorded
> message saying that the number is not in service...
>
>
>
>
> On Fri, Dec 5, 2008 at 1:30 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> What is the hangup cause?
>>
>> /b
>>
>> On Dec 5, 2008, at 10:23 AM, mehdix wrote:
>>
>> > Any Ideas?
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Dec  5 11:08:56 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 11:08:56 -0800
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
	<6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
Message-ID: <87f2f3b90812051108kcd34b5er4fc99e1f7e310aa2@mail.gmail.com>

Doh! Brian is way ahead of me, as usual...

On Fri, Dec 5, 2008 at 11:05 AM, Brian West <brian at freeswitch.org> wrote:
> But you don't see the invite hitting FreeSWITCH?  And you're behind
> NAT?  Make it register every 30 seconds instead of the default 3600
>
> /b
>
> On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:
>
>> Actually, i did not mean that the line is dropped during a call...
>> FS is configured to accept calls from the Junction Networks SIP
>> trunk to make an audio conference.
>> When I start FS and I dial the number all is working fine. But, if I
>> wait for a couple of minutes and then make my call I get an error
>> recorded message saying that the number is not in service...
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mehdi.chaabouni at gmail.com  Fri Dec  5 11:27:59 2008
From: mehdi.chaabouni at gmail.com (MEHDi CHAABOUNi)
Date: Fri, 5 Dec 2008 14:27:59 -0500
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
	<6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
Message-ID: <c9080b60812051127y56195007lad0ecfa3276c0dd1@mail.gmail.com>

I changed the parameter expire-seconds to 30. Now, I'm starting to see the
register request in the console.
I'll wait a couple of hours and get back to you guys.

Thanks

On Fri, Dec 5, 2008 at 2:05 PM, Brian West <brian at freeswitch.org> wrote:

> But you don't see the invite hitting FreeSWITCH?  And you're behind
> NAT?  Make it register every 30 seconds instead of the default 3600
>
> /b
>
> On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:
>
> > Actually, i did not mean that the line is dropped during a call...
> > FS is configured to accept calls from the Junction Networks SIP
> > trunk to make an audio conference.
> > When I start FS and I dial the number all is working fine. But, if I
> > wait for a couple of minutes and then make my call I get an error
> > recorded message saying that the number is not in service...
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jbr at consiglia.dk  Fri Dec  5 11:51:57 2008
From: jbr at consiglia.dk (Jon Bruel)
Date: Fri, 5 Dec 2008 20:51:57 +0100
Subject: [Freeswitch-users] How to force FS using the hosts entry in
	outgoing gateway calls
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A81417D@SBS2008SERVER.consiglia.local>

Thanks Anthony. Using the parameters: 

<param name="proxy" value="domain name"/>
<param name="register-proxy" value="ip address"/>

Returns error 900, and a 'ngrep port port-number' indicates that its doesn't try to register at all. I have now let the server look at a local DNS where I have added a "wrong" A-record. That solves the issue, but your solution would be cleaner. The version is: trunk 10220. /Jon



From anthony.minessale at gmail.com  Fri Dec  5 11:57:29 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 13:57:29 -0600
Subject: [Freeswitch-users] How to force FS using the hosts entry in
	outgoing gateway calls
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A81417D@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A81417D@SBS2008SERVER.consiglia.local>
Message-ID: <191c3a030812051157q2116c468j49516416db4978f7@mail.gmail.com>

you have an older revision.....

put sip:<ip>  instead of just <ip>

I recommend you update and either will work.


On Fri, Dec 5, 2008 at 1:51 PM, Jon Bruel <jbr at consiglia.dk> wrote:

> Thanks Anthony. Using the parameters:
>
> <param name="proxy" value="domain name"/>
> <param name="register-proxy" value="ip address"/>
>
> Returns error 900, and a 'ngrep port port-number' indicates that its
> doesn't try to register at all. I have now let the server look at a local
> DNS where I have added a "wrong" A-record. That solves the issue, but your
> solution would be cleaner. The version is: trunk 10220. /Jon
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From frank at impactfax.com  Fri Dec  5 17:09:00 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 20:09:00 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <87f2f3b90812050938s120f801dy29a95d02f601f89a@mail.gmail.com>
Message-ID: <25a201c9573f$3901df70$33014c0a@ws4>

I tried your suggested test.  Here is the business end of the extension
I tried.

        <action application="set" data="DTMF1=false"/>
        <action application="start_dtmf" data="true"/>
        <action application="tone_detect" data="DTMF1 697,1209 rw +10000
set DTMF1=true"/>  <!-- detect DTMF 1 key-->
        <action application="answer"/>
        <action application="sleep" data="10000"/>
        <action application="info"/>
        <action application="hangup"/>

but I always got DTMF1=false in the info dump.
I am using FS 9210

I have tried sending a call from my sip phone connected to an asterisk
server to FS (dial FS).  I also tried a PSTN call coming in on a PRI to
asterisk and then sip over to FS (another dial from asterisk).  In each
case, pressed 1 several times and the tone_detect never triggered.

Ideas?  Am I doing something stupid or is tone_detect not just right
here?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
 
Those might be necessary if your dtmf's are not already in-band.

Here's a sample extension you could try for testing, dialing 9990:

<extension name="dtmf_detect_test">
    <condition field="destination_number" expression="^9990$">
        <action application="answer"/>
        <action application="tone_detect" data="DTMF1 697,1209 rw
+10000 set DTMF1=true"/>  <!-- detect DTMF 1 key-->
        <action application="sleep" data="10000"/>
        <action application="info"/>
        <action application="hangup"/>
    </condition>
</extension>

Give that a try and at least see if you can detect the tones...
-MC

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From msc at freeswitch.org  Fri Dec  5 17:42:27 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 17:42:27 -0800
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <25a201c9573f$3901df70$33014c0a@ws4>
References: <25a201c9573f$3901df70$33014c0a@ws4>
Message-ID: <5EC8A57C-5D9A-437C-9A7E-B87BAF4B752F@freeswitch.org>

That's a pretty old rev. Any chance you could make current?

-MC

Sent from my iPhone

On Dec 5, 2008, at 5:09 PM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> I tried your suggested test.  Here is the business end of the  
> extension
> I tried.
>
>        <action application="set" data="DTMF1=false"/>
>        <action application="start_dtmf" data="true"/>
>        <action application="tone_detect" data="DTMF1 697,1209 rw  
> +10000
> set DTMF1=true"/>  <!-- detect DTMF 1 key-->
>        <action application="answer"/>
>        <action application="sleep" data="10000"/>
>        <action application="info"/>
>        <action application="hangup"/>
>
> but I always got DTMF1=false in the info dump.
> I am using FS 9210
>
> I have tried sending a call from my sip phone connected to an asterisk
> server to FS (dial FS).  I also tried a PSTN call coming in on a PRI  
> to
> asterisk and then sip over to FS (another dial from asterisk).  In  
> each
> case, pressed 1 several times and the tone_detect never triggered.
>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
>
> Those might be necessary if your dtmf's are not already in-band.
>
> Here's a sample extension you could try for testing, dialing 9990:
>
> <extension name="dtmf_detect_test">
>    <condition field="destination_number" expression="^9990$">
>        <action application="answer"/>
>        <action application="tone_detect" data="DTMF1 697,1209 rw
> +10000 set DTMF1=true"/>  <!-- detect DTMF 1 key-->
>        <action application="sleep" data="10000"/>
>        <action application="info"/>
>        <action application="hangup"/>
>    </condition>
> </extension>
>
> Give that a try and at least see if you can detect the tones...
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gilbertandrew at me.com  Fri Dec  5 18:15:08 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Fri, 05 Dec 2008 21:15:08 -0500
Subject: [Freeswitch-users] How to force FS using the hosts entry
	in	outgoing gateway calls
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>
Message-ID: <9E3B7FE2-C459-40F4-A8B5-DBB064564F32@me.com>

Jon,

You should also be able to do a

'order hosts,bind' in /etc/hosts, no????


On Dec 5, 2008, at 11:43 AM, Jon Bruel wrote:

> For the configuration of a gateway I need to use a specific proxy  
> domain name before the server (Covergence SBC with a BroadWorks  
> Application Server behind) accepts calls. The twist is that the  
> right proxy name points the wrong IP-address (the voicemail server  
> for the account). I have tried to overrule this by adding a host  
> entry (Linux). When I ping to the domain name I get the right  
> address (the one from the host table), but the FS uses the address  
> from the DNS lookup, not the address from the host table. What can I  
> do to force the FS using the entry from the host table? Thanks /Jon.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mehdi.chaabouni at gmail.com  Fri Dec  5 18:34:22 2008
From: mehdi.chaabouni at gmail.com (MEHDi CHAABOUNi)
Date: Fri, 5 Dec 2008 21:34:22 -0500
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <c9080b60812051127y56195007lad0ecfa3276c0dd1@mail.gmail.com>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
	<6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
	<c9080b60812051127y56195007lad0ecfa3276c0dd1@mail.gmail.com>
Message-ID: <c9080b60812051834w2cfd78d7ob81fa53b70563181@mail.gmail.com>

It's working... thanks a lot

On Fri, Dec 5, 2008 at 2:27 PM, MEHDi CHAABOUNi
<mehdi.chaabouni at gmail.com>wrote:

> I changed the parameter expire-seconds to 30. Now, I'm starting to see the
> register request in the console.
> I'll wait a couple of hours and get back to you guys.
>
> Thanks
>
>
> On Fri, Dec 5, 2008 at 2:05 PM, Brian West <brian at freeswitch.org> wrote:
>
>> But you don't see the invite hitting FreeSWITCH?  And you're behind
>> NAT?  Make it register every 30 seconds instead of the default 3600
>>
>> /b
>>
>> On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:
>>
>> > Actually, i did not mean that the line is dropped during a call...
>> > FS is configured to accept calls from the Junction Networks SIP
>> > trunk to make an audio conference.
>> > When I start FS and I dial the number all is working fine. But, if I
>> > wait for a couple of minutes and then make my call I get an error
>> > recorded message saying that the number is not in service...
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From frank at impactfax.com  Fri Dec  5 19:01:32 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 22:01:32 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <5EC8A57C-5D9A-437C-9A7E-B87BAF4B752F@freeswitch.org>
Message-ID: <006d01c9574e$f1a70410$33014c0a@ws4>

Also got it on 9579 as well.  

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins
Sent: Friday, December 05, 2008 8:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call

That's a pretty old rev. Any chance you could make current?

-MC

Sent from my iPhone

On Dec 5, 2008, at 5:09 PM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> I tried your suggested test.  Here is the business end of the  
> extension
> I tried.
>
>        <action application="set" data="DTMF1=false"/>
>        <action application="start_dtmf" data="true"/>
>        <action application="tone_detect" data="DTMF1 697,1209 rw  
> +10000
> set DTMF1=true"/>  <!-- detect DTMF 1 key-->
>        <action application="answer"/>
>        <action application="sleep" data="10000"/>
>        <action application="info"/>
>        <action application="hangup"/>
>
> but I always got DTMF1=false in the info dump.
> I am using FS 9210
>
> I have tried sending a call from my sip phone connected to an asterisk
> server to FS (dial FS).  I also tried a PSTN call coming in on a PRI  
> to
> asterisk and then sip over to FS (another dial from asterisk).  In  
> each
> case, pressed 1 several times and the tone_detect never triggered.
>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
>
> Those might be necessary if your dtmf's are not already in-band.
>
> Here's a sample extension you could try for testing, dialing 9990:
>
> <extension name="dtmf_detect_test">
>    <condition field="destination_number" expression="^9990$">
>        <action application="answer"/>
>        <action application="tone_detect" data="DTMF1 697,1209 rw
> +10000 set DTMF1=true"/>  <!-- detect DTMF 1 key-->
>        <action application="sleep" data="10000"/>
>        <action application="info"/>
>        <action application="hangup"/>
>    </condition>
> </extension>
>
> Give that a try and at least see if you can detect the tones...
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From brian at freeswitch.org  Fri Dec  5 22:01:25 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Dec 2008 00:01:25 -0600
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <25a201c9573f$3901df70$33014c0a@ws4>
References: <25a201c9573f$3901df70$33014c0a@ws4>
Message-ID: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>

make current or install current svn on a different box.

/b

On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:

>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?



From pmhshz at gmail.com  Fri Dec  5 22:42:52 2008
From: pmhshz at gmail.com (shehzad p)
Date: Fri, 5 Dec 2008 22:42:52 -0800 (PST)
Subject: [Freeswitch-users] How to setup TLS between two Freeswitch
 servers
In-Reply-To: <7D08D1D5-5FA8-40FD-BF82-EA9412F6E0D2@freeswitch.org>
References: <20856369.post@talk.nabble.com>
	<7D08D1D5-5FA8-40FD-BF82-EA9412F6E0D2@freeswitch.org>
Message-ID: <20867323.post@talk.nabble.com>


thanks Brian,

thank you so much for useful reply, It works very well now :)...

-
msp


Brian West-3 wrote:
> 
> You would use something like this sofia/profile/ 
> user at remotefsip;transport=tls
> 
> /b
> 
> On Dec 5, 2008, at 9:31 AM, shehzad p wrote:
> 
>>
>>
>> I am wondering how to setup two freeswitch servers to route call  
>> with TLS
>> configured between them.
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/How-to-setup-TLS-between-two-Freeswitch-servers-tp20856369p20867323.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From faisalmaqsoodi at yahoo.com  Fri Dec  5 23:42:43 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Fri, 5 Dec 2008 23:42:43 -0800 (PST)
Subject: [Freeswitch-users] Handling directory of sound files
In-Reply-To: <B25E3A72-98AE-4013-AAE0-CE978FD28717@freeswitch.org>
Message-ID: <427787.87769.qm@web30703.mail.mud.yahoo.com>

Thank u so much. mod local stream really works to play sound files from a local directory sequentially. Now can i jump to a specific file skipping the others? What should i use in dialplan? Have u any idea. Nothing is mentioned in the doc of mod lacal stream about that. 
?????????????????????????????????????????????????????? Faisal

--- On Fri, 12/5/08, Michael S Collins <msc at freeswitch.org> wrote:
From: Michael S Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Handling directory of sound files
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Date: Friday, December 5, 2008, 7:08 AM

Check out mod_localstream on the wiki and see if that sounds like what you need. I'm still learning it all myself but I believe that's where you should start. Please report back with any questions and we will take it from there!
-MC


On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi <faisalmaqsoodi at yahoo.com> wrote:

Its not without music on hold completely. Say, e.g, moh is being played
but when i press 1 it should start playing files contained in a
specific directory sequentially or randomly. I havent got any solution to this problem yet. Can anyone plz guide me to some documentation or anything else regarding this matter.
? ? ? ? ? ? ? ? ? ? ? ?? ???????????????????? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? Faisal


      _______________________________________________
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From faisalmaqsoodi at yahoo.com  Sat Dec  6 01:59:36 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sat, 6 Dec 2008 01:59:36 -0800 (PST)
Subject: [Freeswitch-users] Playing a file again and again
Message-ID: <202061.31652.qm@web30702.mail.mud.yahoo.com>

Hi,
??? Is there any built-in function, like playback, which plays a file again and again unless interrupted. I want to use a simple function not FIFO.
????????????????????????????????????????????????????????????????????????????? Faisal



      
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From odermann at googlemail.com  Sat Dec  6 02:13:01 2008
From: odermann at googlemail.com (Dennis)
Date: Sat, 6 Dec 2008 11:13:01 +0100
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <202061.31652.qm@web30702.mail.mud.yahoo.com>
References: <202061.31652.qm@web30702.mail.mud.yahoo.com>
Message-ID: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>

with sendmsg playback send: loops: -1

2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
> Hi,
>     Is there any built-in function, like playback, which plays a file again
> and again unless interrupted. I want to use a simple function not FIFO.
>
> Faisal
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From faisalmaqsoodi at yahoo.com  Sat Dec  6 06:50:55 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sat, 6 Dec 2008 06:50:55 -0800 (PST)
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>
Message-ID: <297073.11382.qm@web30701.mail.mud.yahoo.com>

I need some more help. I used send msg this way. Is there anything missing bcoz its not working. Plz let me know what else should i do. When i dial 2003, file is not played.
<extension name="Music">
sendmsg 2003
call-command: execute
execute-app-name: playback
execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav
loops: -1
? </extension>

--- On Sat, 12/6/08, Dennis <odermann at googlemail.com> wrote:
From: Dennis <odermann at googlemail.com>
Subject: Re: [Freeswitch-users] Playing a file again and again
To: freeswitch-users at lists.freeswitch.orgi
Date: Saturday, December 6, 2008, 2:13 AM

with sendmsg playback send: loops: -1

2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
> Hi,
>     Is there any built-in function, like playback, which plays a file
again
> and again unless interrupted. I want to use a simple function not FIFO.
>
> Faisal
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From per_moeller at mac.com  Sat Dec  6 06:00:52 2008
From: per_moeller at mac.com (=?iso-8859-1?Q?Per_M=F8ller?=)
Date: Sat, 06 Dec 2008 15:00:52 +0100
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <191c3a030812021205r619ad735le129731ccb8f69d0@mail.gmail.com>
References: <000001c9530d$912d86d0$b3889470$@com>
	<5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
	<000f01c954b4$a616fa60$f244ef20$@com>
	<6E8D2069C08AA84A83D336E996AE4C6702336DC03C@mse17be1.mse17.exchange.ms>
	<191c3a030812021205r619ad735le129731ccb8f69d0@mail.gmail.com>
Message-ID: <002801c957ab$0e00b000$2a021000$@com>

No, only using a single local ip, no stun anywhere.

But I have found the time consuming application, it's DB.

When I commented out the following lines in the default dialplan, that a
call to a local extension would run through, there was no delay:

<action application="db"
data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
<action application="db"
data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_numb
er}"/>
<action application="db"
data="insert/${domain_name}-last_dial/global/${uuid}"/>
<anti-action application="db"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb
er}"/>
<anti-action application="db"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<anti-action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>

A cautious assumption would be that sqllite does not perform as well on
Win32. However I should mention it is compiled as debug, as I cannot get a
release version to compile.

// Per

Fra: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Anthony
Minessale
Sendt: 2. december 2008 21:05
Til: freeswitch-users at lists.freeswitch.org
Emne: Re: [Freeswitch-users] Windows is slow?

is it stun timeout ?

do you have one of the ip set to stun:foo ?

On Tue, Dec 2, 2008 at 1:33 PM, Michael Giagnocavo <mgg at giagnocavo.net>
wrote:
Can you do a console loglevel debug, then send all the output around that
time?

Apart from that, the quickest way might just to attach a debugger, then
break all when it pauses and see where the threads are :).

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Per
M?ller
Sent: Tuesday, December 02, 2008 12:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Windows is slow?

I checked out the trunk version, and it's still slow. However I found one
improvement - it does not crash on shutdown anymore.

Could anymore give me some pointers on how to try to debug this on the
Windows platform?


// Per

Fra: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Carlos
Talbot
Sendt: 2. december 2008 17:13
Til: freeswitch-users at lists.freeswitch.org
Emne: Re: [Freeswitch-users] Windows is slow?

Have you tried the latest msi build? It's based off svn 10564.

Carlos

On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:
I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
on a Windows Vista machine using the precompiled .msi - actually the same
machine).

Using the default configuration files, and using 2 Snom 360 phones I dialed
from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
instantly, but under Windows it takes 1-2 seconds before it starts ringing.

It seems to be in the dialplan the time is spent. From the time I see this
line on the console:

[INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
context default

Until the next thing happens it always takes at least 1 full second, but on
the Mac it happens instantly.

Why is the Windows build this much slower? Is it a known problem?

I get the feeling that the majority of the FS community is Unix based, which
is fine by me, but I would really like to know just how well supported and
stable the Win32 build is and if this is currently a viable way to go, or if
I should stick to Linux/BSD/Mac for production use?


// Per


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_______________________________________________
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_______________________________________________
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Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



From odermann at googlemail.com  Sat Dec  6 08:13:17 2008
From: odermann at googlemail.com (Dennis)
Date: Sat, 6 Dec 2008 17:13:17 +0100
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <297073.11382.qm@web30701.mail.mud.yahoo.com>
References: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>
	<297073.11382.qm@web30701.mail.mud.yahoo.com>
Message-ID: <5e414ed0812060813k4877cc1anbae6a6fc2df9e4ba@mail.gmail.com>

although i do everything with socket outbound and php and not with the
xml-dialplans, i feel, that you are missing the basics.

first you should find out how to make a basic dialplan and how fs is
working. then you can start with playing soundfiles and then, how to
do a looped playback and other nice built in features.
there are lots of dialplan samples delivered with fs and the wiki will
help you to start with the rest.

dennis

2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
> I need some more help. I used send msg this way. Is there anything missing
> bcoz its not working. Plz let me know what else should i do. When i dial
> 2003, file is not played.
> <extension name="Music">
> sendmsg 2003
> call-command: execute
> execute-app-name: playback
> execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav
> loops: -1
>   </extension>
>
> --- On Sat, 12/6/08, Dennis <odermann at googlemail.com> wrote:
>
> From: Dennis <odermann at googlemail.com>
> Subject: Re: [Freeswitch-users] Playing a file again and again
> To: freeswitch-users at lists.freeswitch.orgi
> Date: Saturday, December 6, 2008, 2:13 AM
>
> with sendmsg playback send: loops: -1
>
> 2008/12/6 Faisal Maqsoodi
>  <faisalmaqsoodi at yahoo.com>:
>> Hi,
>>     Is there any built-in function, like playback, which plays a file
> again
>> and again unless interrupted. I want to use a simple function not FIFO.
>>
>> Faisal
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Sat Dec  6 08:37:22 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 6 Dec 2008 08:37:22 -0800
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <5e414ed0812060813k4877cc1anbae6a6fc2df9e4ba@mail.gmail.com>
References: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>
	<297073.11382.qm@web30701.mail.mud.yahoo.com>
	<5e414ed0812060813k4877cc1anbae6a6fc2df9e4ba@mail.gmail.com>
Message-ID: <09F35DA5-6613-4E98-A7D2-960BC24468C3@freeswitch.org>

Faisal,
Dennis makes a good point: you are mixing event socket syntax with  
dialplan syntax. I recommend starting with the dialplan example on the  
wiki. To get a single sound file to play over and over put it in a  
directory by itself.

Also, there is an undocumented feature called ".loc files" that I am  
researching. I believe this feature might give you more options. As  
soon as I can test it and document it I will report back. If any  
community members are using .loc files please let me know how they are  
working for you.

Thanks,
MC

Sent from my iPhone

On Dec 6, 2008, at 8:13 AM, Dennis <odermann at googlemail.com> wrote:

> although i do everything with socket outbound and php and not with the
> xml-dialplans, i feel, that you are missing the basics.
>
> first you should find out how to make a basic dialplan and how fs is
> working. then you can start with playing soundfiles and then, how to
> do a looped playback and other nice built in features.
> there are lots of dialplan samples delivered with fs and the wiki will
> help you to start with the rest.
>
> dennis
>
> 2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
>> I need some more help. I used send msg this way. Is there anything  
>> missing
>> bcoz its not working. Plz let me know what else should i do. When i  
>> dial
>> 2003, file is not played.
>> <extension name="Music">
>> sendmsg 2003
>> call-command: execute
>> execute-app-name: playback
>> execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav
>> loops: -1
>>  </extension>
>>
>> --- On Sat, 12/6/08, Dennis <odermann at googlemail.com> wrote:
>>
>> From: Dennis <odermann at googlemail.com>
>> Subject: Re: [Freeswitch-users] Playing a file again and again
>> To: freeswitch-users at lists.freeswitch.orgi
>> Date: Saturday, December 6, 2008, 2:13 AM
>>
>> with sendmsg playback send: loops: -1
>>
>> 2008/12/6 Faisal Maqsoodi
>> <faisalmaqsoodi at yahoo.com>:
>>> Hi,
>>>    Is there any built-in function, like playback, which plays a file
>> again
>>> and again unless interrupted. I want to use a simple function not  
>>> FIFO.
>>>
>>> Faisal
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Sat Dec  6 08:54:18 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Dec 2008 10:54:18 -0600
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <09F35DA5-6613-4E98-A7D2-960BC24468C3@freeswitch.org>
References: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>
	<297073.11382.qm@web30701.mail.mud.yahoo.com>
	<5e414ed0812060813k4877cc1anbae6a6fc2df9e4ba@mail.gmail.com>
	<09F35DA5-6613-4E98-A7D2-960BC24468C3@freeswitch.org>
Message-ID: <191c3a030812060854k17f59747y6e6e52e4b14175d8@mail.gmail.com>

you can set up an instance of mod_local_stream to endlessly play back a file
and many channels
can listen to it at the same time.

see ext 9999 in the default config.


On Sat, Dec 6, 2008 at 10:37 AM, Michael S Collins <msc at freeswitch.org>wrote:

> Faisal,
> Dennis makes a good point: you are mixing event socket syntax with
> dialplan syntax. I recommend starting with the dialplan example on the
> wiki. To get a single sound file to play over and over put it in a
> directory by itself.
>
> Also, there is an undocumented feature called ".loc files" that I am
> researching. I believe this feature might give you more options. As
> soon as I can test it and document it I will report back. If any
> community members are using .loc files please let me know how they are
> working for you.
>
> Thanks,
> MC
>
> Sent from my iPhone
>
> On Dec 6, 2008, at 8:13 AM, Dennis <odermann at googlemail.com> wrote:
>
> > although i do everything with socket outbound and php and not with the
> > xml-dialplans, i feel, that you are missing the basics.
> >
> > first you should find out how to make a basic dialplan and how fs is
> > working. then you can start with playing soundfiles and then, how to
> > do a looped playback and other nice built in features.
> > there are lots of dialplan samples delivered with fs and the wiki will
> > help you to start with the rest.
> >
> > dennis
> >
> > 2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
> >> I need some more help. I used send msg this way. Is there anything
> >> missing
> >> bcoz its not working. Plz let me know what else should i do. When i
> >> dial
> >> 2003, file is not played.
> >> <extension name="Music">
> >> sendmsg 2003
> >> call-command: execute
> >> execute-app-name: playback
> >> execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav
> >> loops: -1
> >>  </extension>
> >>
> >> --- On Sat, 12/6/08, Dennis <odermann at googlemail.com> wrote:
> >>
> >> From: Dennis <odermann at googlemail.com>
> >> Subject: Re: [Freeswitch-users] Playing a file again and again
> >> To: freeswitch-users at lists.freeswitch.orgi
> >> Date: Saturday, December 6, 2008, 2:13 AM
> >>
> >> with sendmsg playback send: loops: -1
> >>
> >> 2008/12/6 Faisal Maqsoodi
> >> <faisalmaqsoodi at yahoo.com>:
> >>> Hi,
> >>>    Is there any built-in function, like playback, which plays a file
> >> again
> >>> and again unless interrupted. I want to use a simple function not
> >>> FIFO.
> >>>
> >>> Faisal
> >>>
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From frank at impactfax.com  Sat Dec  6 10:19:33 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 6 Dec 2008 13:19:33 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>
Message-ID: <04ab01c957cf$308e78e0$33014c0a@ws4>

Same thing with version 10640 build.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Saturday, December 06, 2008 1:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call

make current or install current svn on a different box.

/b

On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:

>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From uv at yuvalhertzog.com  Sat Dec  6 22:15:19 2008
From: uv at yuvalhertzog.com (UV)
Date: Sun, 7 Dec 2008 17:15:19 +1100
Subject: [Freeswitch-users] VoIP Product/Service and Man/Woman of the year
	2008
Message-ID: <F20429BF4E384EE692D8EBA33A398C9C@UVix>

Biz-news.com
<http://www.linkedin.com/redirect?url=http%3A%2F%2FBiz-news%2Ecom&urlhash=oN
Bl&_t=disc_detail_link>  is conducting a survey to discover the leading VoIP
man/woman of the year 2008, and the best VoIP product or service also for
the year 2008 . The winner will be selected based on the opinions of
professionals and technology enthusiasts in the industry. 

A) best VoIP product or service of 2008 
form: http://voip.biz-news.com/forms/py2008
<http://www.linkedin.com/redirect?url=http%3A%2F%2Fvoip%2Ebiz-news%2Ecom%2Ff
orms%2Fpy2008&urlhash=dspi&_t=disc_detail_link>  

B) VoIP Man/Woman of the year 2008 
form at: http://voip.biz-news.com/forms/my2008
<http://www.linkedin.com/redirect?url=http%3A%2F%2Fvoip%2Ebiz-news%2Ecom%2Ff
orms%2Fmy2008&urlhash=eDrg&_t=disc_detail_link>  

The results will be published early 2009 and share the raw data with the
community.

 

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From astmac at stillnewt.org  Sat Dec  6 22:49:44 2008
From: astmac at stillnewt.org (Martin Joseph)
Date: Sat, 6 Dec 2008 22:49:44 -0800
Subject: [Freeswitch-users] rootkit?
Message-ID: <723A7F88-B907-4C87-97E5-656D4F520272@stillnewt.org>

What is the rootkit item that appears to be added to the SVN of trunk?

Thanks,
Marty



From krice at suspicious.org  Sat Dec  6 22:59:14 2008
From: krice at suspicious.org (Ken Rice)
Date: Sun, 07 Dec 2008 00:59:14 -0600
Subject: [Freeswitch-users] rootkit?
In-Reply-To: <723A7F88-B907-4C87-97E5-656D4F520272@stillnewt.org>
Message-ID: <C560D062.4AE6A%krice@suspicious.org>

Its not really a rootkit... Its installs some screen and emacs profiles that
the FreeSwitch Dev Team use all the time...


> From: Martin Joseph <astmac at stillnewt.org>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 6 Dec 2008 22:49:44 -0800
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] rootkit?
> 
> What is the rootkit item that appears to be added to the SVN of trunk?
> 
> Thanks,
> Marty
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jan.kubr at gmail.com  Sun Dec  7 03:31:26 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Sun, 7 Dec 2008 12:31:26 +0100
Subject: [Freeswitch-users] Read app ignores custom variable when called via
	socket interface
Message-ID: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>

Hi,
I checked out the current trunk (rev 10641) and found out that the
read app ignores the varname parameter, it always puts the result in
the DTMF-Digit variable. I'm calling it via the socket interface:

sendmsg
call-command: execute
execute-app-name: read
execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 #
event-lock:true


In the XML dialplan it works fine:

<action application="read" data="1 1
sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
<action application="phrase" data="spell,${res}"/>


I have been using the socket call above successfully in the 1.0.1 release.

Any ideas? Thanks,

Jan Kubr


From anthony.minessale at gmail.com  Sun Dec  7 11:44:03 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 7 Dec 2008 13:44:03 -0600
Subject: [Freeswitch-users] Read app ignores custom variable when called
	via socket interface
In-Reply-To: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
References: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
Message-ID: <191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>

That seems unlikely.
You sure about that?

The var param is in the middle of the data which is passed as 1 giant string
to the same exact app execution code.
I don't see how it could differentiate

did you try executing the info app right after to see all the vars.

I'm not saying i don't believe you but it seems fishy.



On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr <jan.kubr at gmail.com> wrote:

> Hi,
> I checked out the current trunk (rev 10641) and found out that the
> read app ignores the varname parameter, it always puts the result in
> the DTMF-Digit variable. I'm calling it via the socket interface:
>
> sendmsg
> call-command: execute
> execute-app-name: read
> execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 #
> event-lock:true
>
>
> In the XML dialplan it works fine:
>
> <action application="read" data="1 1
> sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
> <action application="phrase" data="spell,${res}"/>
>
>
> I have been using the socket call above successfully in the 1.0.1 release.
>
> Any ideas? Thanks,
>
> Jan Kubr
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Sun Dec  7 12:28:28 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 7 Dec 2008 14:28:28 -0600
Subject: [Freeswitch-users] Handling directory of sound files
In-Reply-To: <427787.87769.qm@web30703.mail.mud.yahoo.com>
References: <B25E3A72-98AE-4013-AAE0-CE978FD28717@freeswitch.org>
	<427787.87769.qm@web30703.mail.mud.yahoo.com>
Message-ID: <191c3a030812071228recd261eod1146fa3e731e613@mail.gmail.com>

mod_localstream is meant to be a moh source.
Playing a file endlessly can be also just done with extension logic

Make an extension that calls playback then transfers the call back to the
same extension.


On Sat, Dec 6, 2008 at 1:42 AM, Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>wrote:

> Thank u so much. mod local stream really works to play sound files from a
> local directory sequentially. Now can i jump to a specific file skipping the
> others? What should i use in dialplan? Have u any idea. Nothing is mentioned
> in the doc of mod lacal stream about that.
>                                                        Faisal
>
> --- On *Fri, 12/5/08, Michael S Collins <msc at freeswitch.org>* wrote:
>
> From: Michael S Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] Handling directory of sound files
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Date: Friday, December 5, 2008, 7:08 AM
>
> Check out mod_localstream on the wiki and see if that sounds like what you
> need. I'm still learning it all myself but I believe that's where you should
> start. Please report back with any questions and we will take it from there!
>
> -MC
>
>
> On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi < <faisalmaqsoodi at yahoo.com>
> faisalmaqsoodi at yahoo.com> wrote:
>
> Its not without music on hold completely. Say, e.g, moh is being played but
> when i press 1 it should start playing files contained in a specific
> directory sequentially or randomly. I havent got any solution to this
> problem yet. Can anyone plz guide me to some documentation or anything else
> regarding this matter.
>
>                                      Faisal
>
> _______________________________________________
> Freeswitch-users mailing list
> <Freeswitch-users at lists.freeswitch.org>
> Freeswitch-users at lists.freeswitch.org
> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users><http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> <http://www.freeswitch.org>http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From odermann at googlemail.com  Mon Dec  8 01:28:08 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 10:28:08 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
Message-ID: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>

hi,

we are fighting with two flaws in fs and would be happy, if they could
be fixed. we are using socket outbound.

1.) hangup a call in ringing state:
this worked in one of the last fs versions, but suddenly does not work anymore.

let's say, we have an inbound call and do 3 originates to different
targets. all 3 targets are in ringing state. the target, which answers
first, will be bridged with the inbound call, the other two (still
ringing) targets should be hung up.

we do not want fs to hang up the other two originates automatically.
we want to hang up the other two originates by sending the hangups. we
set "hangup_after_bridge=false" and "park_after_bridge=true".
we do sendmsg hangup uuid. but the originates are first hung up, when
they are answered. when they are in ringing state, the hangup will do
nothing (anymore).

as i said, it worked before, so i assume, that something has changed
in the latest trunks.



2.) sendmsg uuid *whatever* can cause to excute the command on the wrong uuid:

let's say, we have an inbound call and an outbound call - at least we
thing we have it ;-).
now we do for example sendmsg outbound_uuid hangup to hangup the
outbound call. but, if the uuid of the outbound call does not exist
(because there was a problem or something), the inbound will be hung
up instead.
the same happens with all sent messages to an uuid, which does not
exist. if we want to do a playback for the same outbound, the inbound
will hear it, if the outbound_uuid does not exist.

perhaps this is a feature, but i think that it would be nicer and more
reliable, if the sendmsg is only executed on the given uuid. if the
given uuid does not exist, nothing should happen or even nicer, an
event with an error should be sent to the socket.


thanks
dennis


From odermann at googlemail.com  Mon Dec  8 01:47:46 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 10:47:46 +0100
Subject: [Freeswitch-users] Two nice to have features in fs
Message-ID: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>

hi,

we are using socket outbound and found out, that there are two
(perhaps) small things, we would like to see in fs.

1.) if we try to playback a soundfile, which does not exist, we do not
get an error or something. in the cli of fs we can see the error, but
we do not get anything about it over the socket.

we get an execute event for the playback and directly after it an
execute-complete.
wouldn't it be nicer, if one could get a notice about an error, like:
execute-complete-error or a variable, which is set with an error, if
an error occours?


2.) the session heartbeat event is an absolutely great feature in fs.
but sometimes we would like to get the session heartbeat every 5
seconds. at the moment the allowed minimum is every 10 seconds. a
smaller setting, like every 5 seconds, will result in a session
heartbeat of every 60 seconds.

to help us out, we edit the switch_core_session.c at line 899 and
change the "seconds < 10" to "seconds < 5".

because one has to set "enable_heartbeat_events=5" manually, i do not
think, that there is a risk, that others, who do not want the
heartbeat to come that often, will be negatively affected by this
change.

might it be possible, to do the same changes to the default code?


thanks
dennis


From regs at kinetix.gr  Mon Dec  8 02:11:32 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Mon, 08 Dec 2008 12:11:32 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
Message-ID: <493CF2D4.6010904@kinetix.gr>

Any updates about the "which side hanged up" potential variable?

Michael S Collins wrote:
> Makes sense. I will look into this. 
> -MC
>
>
> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr 
> <mailto:regs at kinetix.gr>> wrote:
>
>> I am sending this second email to request/suggest/enquire about 
>> something relevant :
>>
>> Wouldn't it be useful to know which end of a specific call leg send 
>> the protocol
>> specific hangup cause? Otherwise it would be difficult to understand 
>> what really happened.
>>
>>
>>
>> Michael S Collins wrote:
>>> I will do some research on this and let you know what I find out.  
>>> Question: are these internal calls or pstn or ?? Just curious about  
>>> your environment.
>>>
>>> Thanks,
>>> MC
>>>
>>>
>>>
>>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr <mailto:regs at kinetix.gr>>  
>>> wrote:
>>>
>>>   
>>>> The proto_specific_hangup_cause is missing on one of the two
>>>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>> When the callee hangs up it is missing from the b-leg CDR. Is this  
>>>> nornal?
>>>>
>>>> And another question : what piece of info could inform me about who
>>>> hanged up?
>>>>
>>>>
>>>> -- 
>>>> -------------------------------------------
>>>> Apostolos Pantsiopoulos
>>>> Kinetix Tele.com R & D
>>>> email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>> -------------------------------------------
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>     
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>   
>>
>>
>> -- 
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>> ------------------------------------------- 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org 
>> <mailto:Freeswitch-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From jonas.gauffin at gmail.com  Mon Dec  8 02:36:26 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 8 Dec 2008 11:36:26 +0100
Subject: [Freeswitch-users] Autoanswer
In-Reply-To: <d021d3be0811260827o68efc7drf2985ad684c72b71@mail.gmail.com>
References: <d021d3be0811260720p29736159p953b6bdd5bef6621@mail.gmail.com>
	<191c3a030811260819l5081b3a2q9b606d7a109d58be@mail.gmail.com>
	<d021d3be0811260827o68efc7drf2985ad684c72b71@mail.gmail.com>
Message-ID: <d021d3be0812080236v4c7bb4b4he8ab7faae02c09f0@mail.gmail.com>

I've tried what you said, and both legs still get auto answered.

My string:
api originate
{gate_user_id=157,gate_site_id=87,origination_caller_id_name='Namie
Amuro'}[sip_invite_params=intercom=true,sip_h_Call-Info=<sip:$${domain}>;answer-after=0,sip_auto_answer=true]sofia/localdomain/u1000157%
192.168.1.111 '&execute_extension(1202 XML)'


On Wed, Nov 26, 2008 at 5:27 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Ahh. gr8. thanks.
>
>
> On Wed, Nov 26, 2008 at 5:19 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you can put vars in [ ] before each channel in the list to apply them only
>> to that one channel.
>>
>> {global=true}[only_this_channel=true]sofia/foo/foo at bar.com
>> ,[only_this_channel_again=true]sofia/foo/baz at bar.com
>>
>>
>> On Wed, Nov 26, 2008 at 9:20 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello
>>> I send an API command through the event socket that looks like this (the
>>> first two variables are used by our server):
>>>
>>> api originate
>>> {gate_user_id=44,gate_site_id=1,sip_invite_params=intercom=true,sip_h_Call-Info=<sip:$${domain}>;answer-after=0,sip_auto_answer=true,origination_caller_id_name='Jonas
>>> Gauffin',origination_caller_id_number=+4623666XXXX,sip_auto_answer=true}sofia/localdomain/u1000044%
>>> 192.168.1.112 '&execute_extension(8901 XML)'
>>>
>>> The command works just as it should. The problem is that the auto-answer
>>> variables seems to stick to the b-leg (execute extensions), which means that
>>> both calls gets auto answered. What I want is that only the first call gets
>>> answered.
>>>
>>> (A logged in user presses the "call" icon in our webdirectory, which
>>> makes freeswitch use his phone to call 8901)
>>>
>>> //Jonas
>>>
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jan.kubr at gmail.com  Mon Dec  8 02:48:30 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Mon, 8 Dec 2008 11:48:30 +0100
Subject: [Freeswitch-users] Read app ignores custom variable when called
	via socket interface
In-Reply-To: <191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>
References: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
	<191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>
Message-ID: <698401620812080248u5f866d7es950d0019686b2c99@mail.gmail.com>

OK my bad. The variable is set (I can see it in the Freeswitch console
when I use the info app), but they are only not send to me via the
socket interface. I get the "variable_*" variables only in the
beginning (after calling connect), but not in the events. How do I
enable this?
Thanks,
Jan

On Sun, Dec 7, 2008 at 8:44 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> That seems unlikely.
> You sure about that?
>
> The var param is in the middle of the data which is passed as 1 giant string
> to the same exact app execution code.
> I don't see how it could differentiate
>
> did you try executing the info app right after to see all the vars.
>
> I'm not saying i don't believe you but it seems fishy.
>
>
>
> On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr <jan.kubr at gmail.com> wrote:
>>
>> Hi,
>> I checked out the current trunk (rev 10641) and found out that the
>> read app ignores the varname parameter, it always puts the result in
>> the DTMF-Digit variable. I'm calling it via the socket interface:
>>
>> sendmsg
>> call-command: execute
>> execute-app-name: read
>> execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 #
>> event-lock:true
>>
>>
>> In the XML dialplan it works fine:
>>
>> <action application="read" data="1 1
>> sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
>> <action application="phrase" data="spell,${res}"/>
>>
>>
>> I have been using the socket call above successfully in the 1.0.1 release.
>>
>> Any ideas? Thanks,
>>
>> Jan Kubr
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From carole.olivier at enst.fr  Mon Dec  8 05:10:16 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Mon, 8 Dec 2008 05:10:16 -0800 (PST)
Subject: [Freeswitch-users] conference configured to call automatically
 the attended does not work
In-Reply-To: <20856465.post@talk.nabble.com>
References: <20856465.post@talk.nabble.com>
Message-ID: <20895038.post@talk.nabble.com>


I have found my mistake.

In the dialplan I have written <action
application="conference_set_auto_outcall" data="sofia/internal/.../> instead
of <action application="conference_set_auto_outcall" data="user/.../> 

Sorry for the post,
Carole


Carole O. wrote:
> 
> Hello,
> 
> I have got some problems for the configuration of a simple conference
> which should be established by calling an extension and automatically
> inviting 2 people.
> 
> Actually, this is based on the default configuration of Freeswitch
> (extension 0911). I have changed it a little:
> 
> <extension name="conference_test">
>     <condition field="destination_number" expression="^0911$">
>     <!--  These parameters effect the outcalls made once you join 
>       --> 
>       <action application="set"
> data="conference_auto_outcall_caller_id_name=telephoneX" /> 
>       <action application="set"
> data="conference_auto_outcall_id_number=0911" /> 
>       <action application="set" data="conference_auto_outcall_timeout=60"
> /> 
>       <action application="set" data="conference_auto_outcall_flags=none"
> /> 
>  
>     <!--  list of devices called  --> 
>       <action application="conference_set_auto_outcall"
> data="sofia/internal/1010@$${domain}" /> 
>       <action application="conference_set_auto_outcall"
> data="sofia/internal/1002@$${domain}" /> 
> 
>       <action application="conference" data="conferenceTest at default" /> 
>     </condition>
>  </extension>
> 
> 
> I have attached a file with the console errors. There are some errors (moh
> errors) but since these were also present for room conference and it did
> not prevent it for working, I guess this is not the fundamental reason for
> the previous problem.
> 
> I have an additional question. I have installed freeswitch from
> opensuse.org, there is a simple "one-click installation" but I am not sure
> this was a good idea, it seems to be light isn't?
> 
> Thanks for your help,
> Carole
>  http://www.nabble.com/file/p20856465/error_console.txt error_console.txt 
> 

-- 
View this message in context: http://www.nabble.com/conference-configured-to-call-automatically-the-attended-does-not-work-tp20856465p20895038.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From mike at jerris.com  Mon Dec  8 05:33:01 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 8 Dec 2008 08:33:01 -0500
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
Message-ID: <2BA1F89F-C5A8-4DC6-B297-052363CC178B@jerris.com>

Can you please file bugs on http://jira.freeswitch.org with full sip  
trace and FreeSWITCH debug output of these cases.


On Dec 8, 2008, at 4:28 AM, Dennis wrote:

> hi,
>
> we are fighting with two flaws in fs and would be happy, if they could
> be fixed. we are using socket outbound.
>
> 1.) hangup a call in ringing state:
> this worked in one of the last fs versions, but suddenly does not  
> work anymore.
>
> let's say, we have an inbound call and do 3 originates to different
> targets. all 3 targets are in ringing state. the target, which answers
> first, will be bridged with the inbound call, the other two (still
> ringing) targets should be hung up.
>
> we do not want fs to hang up the other two originates automatically.
> we want to hang up the other two originates by sending the hangups. we
> set "hangup_after_bridge=false" and "park_after_bridge=true".
> we do sendmsg hangup uuid. but the originates are first hung up, when
> they are answered. when they are in ringing state, the hangup will do
> nothing (anymore).
>
> as i said, it worked before, so i assume, that something has changed
> in the latest trunks.
>
>
>
> 2.) sendmsg uuid *whatever* can cause to excute the command on the  
> wrong uuid:
>
> let's say, we have an inbound call and an outbound call - at least we
> thing we have it ;-).
> now we do for example sendmsg outbound_uuid hangup to hangup the
> outbound call. but, if the uuid of the outbound call does not exist
> (because there was a problem or something), the inbound will be hung
> up instead.
> the same happens with all sent messages to an uuid, which does not
> exist. if we want to do a playback for the same outbound, the inbound
> will hear it, if the outbound_uuid does not exist.
>
> perhaps this is a feature, but i think that it would be nicer and more
> reliable, if the sendmsg is only executed on the given uuid. if the
> given uuid does not exist, nothing should happen or even nicer, an
> event with an error should be sent to the socket.
>
>
> thanks
> dennis
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Mon Dec  8 05:34:42 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 8 Dec 2008 08:34:42 -0500
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
Message-ID: <1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>


On Dec 8, 2008, at 4:47 AM, Dennis wrote:

> at the moment the allowed minimum is every 10 seconds. a
> smaller setting, like every 5 seconds, will result in a session
> heartbeat of every 60 seconds.

Huh?


From anthony.minessale at gmail.com  Mon Dec  8 05:48:02 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 07:48:02 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493CF2D4.6010904@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
	<493CF2D4.6010904@kinetix.gr>
Message-ID: <191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>

sip_hangup_disposition will be set to recv_bye on the side that was hungup.


On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  Any updates about the "which side hanged up" potential variable?
>
> Michael S Collins wrote:
>
> Makes sense. I will look into this.
> -MC
>
>
> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
> wrote:
>
>  I am sending this second email to request/suggest/enquire about something
> relevant :
>
> Wouldn't it be useful to know which end of a specific call leg send the
> protocol
> specific hangup cause? Otherwise it would be difficult to understand what
> really happened.
>
>
>
> Michael S Collins wrote:
>
> I will do some research on this and let you know what I find out.
> Question: are these internal calls or pstn or ?? Just curious about
> your environment.
>
> Thanks,
> MC
>
>
>
> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <regs at kinetix.gr>regs at kinetix.gr>
> wrote:
>
>
>
>  The proto_specific_hangup_cause is missing on one of the two
> call legs. When the caller hangs up it is missing from the a-leg CDR.
> When the callee hangs up it is missing from the b-leg CDR. Is this
> nornal?
>
> And another question : what piece of info could inform me about who
> hanged up?
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>  _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>   _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec  8 06:04:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 08:04:51 -0600
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
Message-ID: <191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>

#2 was because when you sendmsg with no uuid on an outbound socket it
defaults to the session who called you.
I changed to code to make a distinction between not supplying a uuid and
supplying an invalid uuid.

#1 seems hard to believe.  Please provide a console trace of the channel
*ignoring* the hangup command.


On Mon, Dec 8, 2008 at 3:28 AM, Dennis <odermann at googlemail.com> wrote:

> hi,
>
> we are fighting with two flaws in fs and would be happy, if they could
> be fixed. we are using socket outbound.
>
> 1.) hangup a call in ringing state:
> this worked in one of the last fs versions, but suddenly does not work
> anymore.
>
> let's say, we have an inbound call and do 3 originates to different
> targets. all 3 targets are in ringing state. the target, which answers
> first, will be bridged with the inbound call, the other two (still
> ringing) targets should be hung up.
>
> we do not want fs to hang up the other two originates automatically.
> we want to hang up the other two originates by sending the hangups. we
> set "hangup_after_bridge=false" and "park_after_bridge=true".
> we do sendmsg hangup uuid. but the originates are first hung up, when
> they are answered. when they are in ringing state, the hangup will do
> nothing (anymore).
>
> as i said, it worked before, so i assume, that something has changed
> in the latest trunks.
>
>
>
> 2.) sendmsg uuid *whatever* can cause to excute the command on the wrong
> uuid:
>
> let's say, we have an inbound call and an outbound call - at least we
> thing we have it ;-).
> now we do for example sendmsg outbound_uuid hangup to hangup the
> outbound call. but, if the uuid of the outbound call does not exist
> (because there was a problem or something), the inbound will be hung
> up instead.
> the same happens with all sent messages to an uuid, which does not
> exist. if we want to do a playback for the same outbound, the inbound
> will hear it, if the outbound_uuid does not exist.
>
> perhaps this is a feature, but i think that it would be nicer and more
> reliable, if the sendmsg is only executed on the given uuid. if the
> given uuid does not exist, nothing should happen or even nicer, an
> event with an error should be sent to the socket.
>
>
> thanks
> dennis
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From regs at kinetix.gr  Mon Dec  8 06:13:47 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Mon, 08 Dec 2008 16:13:47 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
Message-ID: <493D2B9B.6050205@kinetix.gr>

Not necessarily. For instance I got a "send_cancel" when the
calling party hanged up before the other party could pick up.
Also, shouldn't something like that be protocol/technology
neutral?



Anthony Minessale wrote:
> sip_hangup_disposition will be set to recv_bye on the side that was 
> hungup.
>
>
> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     Any updates about the "which side hanged up" potential variable?
>
>     Michael S Collins wrote:
>>     Makes sense. I will look into this. 
>>     -MC
>>
>>
>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>>
>>>     I am sending this second email to request/suggest/enquire about
>>>     something relevant :
>>>
>>>     Wouldn't it be useful to know which end of a specific call leg
>>>     send the protocol
>>>     specific hangup cause? Otherwise it would be difficult to
>>>     understand what really happened.
>>>
>>>
>>>
>>>     Michael S Collins wrote:
>>>>     I will do some research on this and let you know what I find out.  
>>>>     Question: are these internal calls or pstn or ?? Just curious about  
>>>>     your environment.
>>>>
>>>>     Thanks,
>>>>     MC
>>>>
>>>>
>>>>
>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <mailto:regs at kinetix.gr>regs at kinetix.gr <mailto:regs at kinetix.gr>>  
>>>>     wrote:
>>>>
>>>>       
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this  
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     -- 
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>         
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>       
>>>
>>>
>>>     -- 
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     ------------------------------------------- 
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org
>>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From steveayre at gmail.com  Mon Dec  8 05:42:14 2008
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 8 Dec 2008 13:42:14 +0000
Subject: [Freeswitch-users] rootkit?
In-Reply-To: <723A7F88-B907-4C87-97E5-656D4F520272@stillnewt.org>
References: <723A7F88-B907-4C87-97E5-656D4F520272@stillnewt.org>
Message-ID: <eafa772f0812080542p417cf9ddu12867963d6892aeb@mail.gmail.com>

A few files which you can choose to install to let the Freeswitch
developers access your machine remotely if you ask them to look at a
problem you're having with Freeswitch.

It's not a real rootkit - just the SSH public key for their private
key (so they don't need a password) and configuration files for bash +
emacs (so they're working in the environment they're used to). It
doesn't burrow into your OS, hide itself or any of the other insidious
things a real rootkit does, and once you delete the files they can no
longer access your machine.

It isn't installed unless you choose to do so.

-Steve



2008/12/7 Martin Joseph <astmac at stillnewt.org>:
> What is the rootkit item that appears to be added to the SVN of trunk?
>
> Thanks,
> Marty
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From odermann at googlemail.com  Mon Dec  8 07:18:18 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 16:18:18 +0100
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
	<1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
Message-ID: <5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>

> Huh?

src/switch_core_session.c vom line 899 to 901:

	if (seconds < 10) {
		seconds = 60;
	}


From anthony.minessale at gmail.com  Mon Dec  8 07:30:15 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 09:30:15 -0600
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
	<1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
	<5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>
Message-ID: <191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>

done

On Mon, Dec 8, 2008 at 9:18 AM, Dennis <odermann at googlemail.com> wrote:

> > Huh?
>
> src/switch_core_session.c vom line 899 to 901:
>
>        if (seconds < 10) {
>                seconds = 60;
>         }
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec  8 07:42:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 09:42:51 -0600
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
	<1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
	<5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>
	<191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>
Message-ID: <191c3a030812080742w7e6e2a3co42453a7b41d88826@mail.gmail.com>

both done

On Mon, Dec 8, 2008 at 9:30 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> done
>
>
> On Mon, Dec 8, 2008 at 9:18 AM, Dennis <odermann at googlemail.com> wrote:
>
>> > Huh?
>>
>> src/switch_core_session.c vom line 899 to 901:
>>
>>        if (seconds < 10) {
>>                seconds = 60;
>>         }
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Mon Dec  8 07:47:38 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 16:47:38 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
Message-ID: <5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>

> #2 was because when you sendmsg with no uuid on an outbound socket it
> defaults to the session who called you.
> I changed to code to make a distinction between not supplying a uuid and
> supplying an invalid uuid.

anthony, thanks for the quick reaction!

we just tested you changes and it works the opposite way it should.

this means: when we do not send an uuid, we get an an error
(Reply-Text => -ERR invalid session id []). if we send a wrong/not
existing uuid, the command will be executed on the inbound uuid.




> #1 seems hard to believe.  Please provide a console trace of the channel
> *ignoring* the hangup command.

i know it is hard to believe, we didn't believe it either ;-)

have a look at http://pastebin.freeswitch.org/6367

what we simply do here: the inbound is coming in, then we do an
originate and hang up the inbound. then we directly send a hangup for
the outbound. the outbound will go on ringing.
then, when the ringing outbound is answered, we directly get the hangup.
fs gets the hangup and remembers it, but seems to wait till the answer
to execute this command.


From anthony.minessale at gmail.com  Mon Dec  8 07:52:17 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 09:52:17 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493D2B9B.6050205@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
	<493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
	<493D2B9B.6050205@kinetix.gr>
Message-ID: <191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>

it is protocol neutral, that's why it starts with sip_

the variable can be any of:

send_bye
recv_bye
send_cancel
send_refuse


using that value you can determine the information you asked.  I answered
your specific question which was:
determining "which side hanged up".  Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.

I already added 2 patches for you right.  Just be clear about what you want.



On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  Not necessarily. For instance I got a "send_cancel" when the
> calling party hanged up before the other party could pick up.
> Also, shouldn't something like that be protocol/technology
> neutral?
>
>
>
> Anthony Minessale wrote:
>
> sip_hangup_disposition will be set to recv_bye on the side that was hungup.
>
>
> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:
>
>>  Any updates about the "which side hanged up" potential variable?
>>
>> Michael S Collins wrote:
>>
>>  Makes sense. I will look into this.
>> -MC
>>
>>
>> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>> wrote:
>>
>>  I am sending this second email to request/suggest/enquire about
>> something relevant :
>>
>> Wouldn't it be useful to know which end of a specific call leg send the
>> protocol
>> specific hangup cause? Otherwise it would be difficult to understand what
>> really happened.
>>
>>
>>
>> Michael S Collins wrote:
>>
>> I will do some research on this and let you know what I find out.
>> Question: are these internal calls or pstn or ?? Just curious about
>> your environment.
>>
>> Thanks,
>> MC
>>
>>
>>
>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <regs at kinetix.gr>regs at kinetix.gr>
>> wrote:
>>
>>
>>
>>  The proto_specific_hangup_cause is missing on one of the two
>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>> When the callee hangs up it is missing from the b-leg CDR. Is this
>> nornal?
>>
>> And another question : what piece of info could inform me about who
>> hanged up?
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>  _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>   _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>  ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Mon Dec  8 08:07:01 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 17:07:01 +0100
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
	<1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
	<5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>
	<191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>
Message-ID: <5e414ed0812080807w609e56f4o3c380f5417e32010@mail.gmail.com>

great, that works! thanks a lot!

just tested the changes according an error, when a file is missing.
thanks again!


2008/12/8 Anthony Minessale <anthony.minessale at gmail.com>:
> done
>
> On Mon, Dec 8, 2008 at 9:18 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> > Huh?
>>
>> src/switch_core_session.c vom line 899 to 901:
>>
>>        if (seconds < 10) {
>>                seconds = 60;
>>        }
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Mon Dec  8 08:08:48 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 10:08:48 -0600
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
Message-ID: <191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>

try the sendmsg issue again

are you doing the hangup with

api uuid_kill <uuid>



On Mon, Dec 8, 2008 at 9:47 AM, Dennis <odermann at googlemail.com> wrote:

> > #2 was because when you sendmsg with no uuid on an outbound socket it
> > defaults to the session who called you.
> > I changed to code to make a distinction between not supplying a uuid and
> > supplying an invalid uuid.
>
> anthony, thanks for the quick reaction!
>
> we just tested you changes and it works the opposite way it should.
>
> this means: when we do not send an uuid, we get an an error
> (Reply-Text => -ERR invalid session id []). if we send a wrong/not
> existing uuid, the command will be executed on the inbound uuid.
>
>
>
>
> > #1 seems hard to believe.  Please provide a console trace of the channel
> > *ignoring* the hangup command.
>
> i know it is hard to believe, we didn't believe it either ;-)
>
> have a look at http://pastebin.freeswitch.org/6367
>
> what we simply do here: the inbound is coming in, then we do an
> originate and hang up the inbound. then we directly send a hangup for
> the outbound. the outbound will go on ringing.
> then, when the ringing outbound is answered, we directly get the hangup.
> fs gets the hangup and remembers it, but seems to wait till the answer
> to execute this command.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Mon Dec  8 08:19:02 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 17:19:02 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
Message-ID: <5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>

i have to shift places. will be back in a few minutes and test.

no, we are using the simple sendmsg uuid hangup. as far as we
remember, we do not use api uuid_kill, because we do not get a hangup
event with this.


2008/12/8 Anthony Minessale <anthony.minessale at gmail.com>:
> try the sendmsg issue again
>
> are you doing the hangup with
>
> api uuid_kill <uuid>


From anthony.minessale at gmail.com  Mon Dec  8 08:44:58 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 10:44:58 -0600
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
Message-ID: <191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>

you would get a hangup event in either case.


On Mon, Dec 8, 2008 at 10:19 AM, Dennis <odermann at googlemail.com> wrote:

> i have to shift places. will be back in a few minutes and test.
>
> no, we are using the simple sendmsg uuid hangup. as far as we
> remember, we do not use api uuid_kill, because we do not get a hangup
> event with this.
>
>
> 2008/12/8 Anthony Minessale <anthony.minessale at gmail.com>:
> > try the sendmsg issue again
> >
> > are you doing the hangup with
> >
> > api uuid_kill <uuid>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Mon Dec  8 08:56:15 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 17:56:15 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
Message-ID: <5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>

> you would get a hangup event in either case.

yes, you are right. we just tested and saw that. the reason for
sendmsg hangup, was the sometimes useful event-lock.

it works with api uuid_kill as we wanted. but with sendmsg hangup it
still does not work. shouldn't sendmsg hangup work like uuid_kill
here? how useful could it be, to let it ring, when the hangup was
already sent and is immediately executed when the anser is sent?


#2 now works perfectly. thanks for the great support!


From mrjoebain at gmail.com  Mon Dec  8 08:57:11 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Mon, 8 Dec 2008 16:57:11 +0000
Subject: [Freeswitch-users] Catching hangups
Message-ID: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>

Hi,

I'm writing an IVR in Lua and am having problems dealing with hangups
cleanly. Very often session:ready() reports true long after I have hung up
and the hangup hook function I have set doesn't get called either. It seems
to report that the session is active indefinitely in some cases where a loop
keeps trying to get some dtmf key presses. Is there any trick to using
session:ready() or the hangup hook that I might have missed?

On a slightly related point I can't seem to access the session properties,
e.g. session.caller_id_num has a value of nil. Any thoughts here?

Thanks in advance,

Joe Bain
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From anthony.minessale at gmail.com  Mon Dec  8 09:11:52 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 11:11:52 -0600
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
	<5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
Message-ID: <191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>

channels in originate were not checking for private events.
now they should but if send them commands to do crazy stuff like play a file
while they are
in the middle of originating there could be ill side effects (e.g. play file
before media was established etc which could cause the call to abort)

btw you can send

call-command: hangup
hangup-cause: normal_clearing

in place of
call-command: execute
execute-app-name: hangup
execute-app-arg: normal_clearing


On Mon, Dec 8, 2008 at 10:56 AM, Dennis <odermann at googlemail.com> wrote:

> > you would get a hangup event in either case.
>
> yes, you are right. we just tested and saw that. the reason for
> sendmsg hangup, was the sometimes useful event-lock.
>
> it works with api uuid_kill as we wanted. but with sendmsg hangup it
> still does not work. shouldn't sendmsg hangup work like uuid_kill
> here? how useful could it be, to let it ring, when the hangup was
> already sent and is immediately executed when the anser is sent?
>
>
> #2 now works perfectly. thanks for the great support!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Mon Dec  8 09:13:34 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Dec 2008 09:13:34 -0800
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
Message-ID: <87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>

Joe,

A few questions... what svn rev are you running? Which operating
system? Finally, is it possible for you to put your dialplan and Lua
script up at pastebin.freeswitch.org?

Thanks,
MC

On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
> Hi,
>
> I'm writing an IVR in Lua and am having problems dealing with hangups
> cleanly. Very often session:ready() reports true long after I have hung up
> and the hangup hook function I have set doesn't get called either. It seems
> to report that the session is active indefinitely in some cases where a loop
> keeps trying to get some dtmf key presses. Is there any trick to using
> session:ready() or the hangup hook that I might have missed?
>
> On a slightly related point I can't seem to access the session properties,
> e.g. session.caller_id_num has a value of nil. Any thoughts here?
>
> Thanks in advance,
>
> Joe Bain
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Mon Dec  8 09:18:01 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 11:18:01 -0600
Subject: [Freeswitch-users] Read app ignores custom variable when called
	via socket interface
In-Reply-To: <698401620812080248u5f866d7es950d0019686b2c99@mail.gmail.com>
References: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
	<191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>
	<698401620812080248u5f866d7es950d0019686b2c99@mail.gmail.com>
Message-ID: <191c3a030812080918x6acd7564q3247055cdc941641@mail.gmail.com>

i added a patch to index the variables on the
SWITCH_EVENT_CHANNEL_EXECUTE_COMPLETE
if you want to update

otherwise you can use uuid_getvar to retrieve the variable


On Mon, Dec 8, 2008 at 4:48 AM, Jan Kubr <jan.kubr at gmail.com> wrote:

> OK my bad. The variable is set (I can see it in the Freeswitch console
> when I use the info app), but they are only not send to me via the
> socket interface. I get the "variable_*" variables only in the
> beginning (after calling connect), but not in the events. How do I
> enable this?
> Thanks,
> Jan
>
> On Sun, Dec 7, 2008 at 8:44 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > That seems unlikely.
> > You sure about that?
> >
> > The var param is in the middle of the data which is passed as 1 giant
> string
> > to the same exact app execution code.
> > I don't see how it could differentiate
> >
> > did you try executing the info app right after to see all the vars.
> >
> > I'm not saying i don't believe you but it seems fishy.
> >
> >
> >
> > On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr <jan.kubr at gmail.com> wrote:
> >>
> >> Hi,
> >> I checked out the current trunk (rev 10641) and found out that the
> >> read app ignores the varname parameter, it always puts the result in
> >> the DTMF-Digit variable. I'm calling it via the socket interface:
> >>
> >> sendmsg
> >> call-command: execute
> >> execute-app-name: read
> >> execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000
> #
> >> event-lock:true
> >>
> >>
> >> In the XML dialplan it works fine:
> >>
> >> <action application="read" data="1 1
> >> sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
> >> <action application="phrase" data="spell,${res}"/>
> >>
> >>
> >> I have been using the socket call above successfully in the 1.0.1
> release.
> >>
> >> Any ideas? Thanks,
> >>
> >> Jan Kubr
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From odermann at googlemail.com  Mon Dec  8 09:26:52 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 18:26:52 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
	<5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
	<191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
Message-ID: <5e414ed0812080926ob7a134emc4a1d00d447f87b8@mail.gmail.com>

thanks, now it works as we expected.

and thanks for the hint, how we should send the hangup with sendmsg.
we will do it your way :-)


2008/12/8 Anthony Minessale <anthony.minessale at gmail.com>:
> channels in originate were not checking for private events.
> now they should but if send them commands to do crazy stuff like play a file
> while they are
> in the middle of originating there could be ill side effects (e.g. play file
> before media was established etc which could cause the call to abort)
>
> btw you can send
>
> call-command: hangup
> hangup-cause: normal_clearing
>
> in place of
> call-command: execute
> execute-app-name: hangup
> execute-app-arg: normal_clearing
>
>
> On Mon, Dec 8, 2008 at 10:56 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> > you would get a hangup event in either case.
>>
>> yes, you are right. we just tested and saw that. the reason for
>> sendmsg hangup, was the sometimes useful event-lock.
>>
>> it works with api uuid_kill as we wanted. but with sendmsg hangup it
>> still does not work. shouldn't sendmsg hangup work like uuid_kill
>> here? how useful could it be, to let it ring, when the hangup was
>> already sent and is immediately executed when the anser is sent?
>>
>>
>> #2 now works perfectly. thanks for the great support!
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dule.maillist at gmail.com  Mon Dec  8 09:51:29 2008
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 8 Dec 2008 12:51:29 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <04ab01c957cf$308e78e0$33014c0a@ws4>
References: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>
	<04ab01c957cf$308e78e0$33014c0a@ws4>
Message-ID: <914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>

>From my understanding, I didn't think tone_detect detects DTMF since it's
dual frequencies, rather tone_detect detects single frequencies like fax
tones.

I would just run an IVR with a session.read or session.getDigits to collect
DTMF.

Dan

On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com> wrote:

> Same thing with version 10640 build.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Saturday, December 06, 2008 1:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
>
> make current or install current svn on a different box.
>
> /b
>
> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>
> >
> > Ideas?  Am I doing something stupid or is tone_detect not just right
> > here?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Dec  8 09:54:40 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 11:54:40 -0600
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>
References: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>
	<04ab01c957cf$308e78e0$33014c0a@ws4>
	<914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>
Message-ID: <191c3a030812080954x6bbf0c85o209d157e6a05412e@mail.gmail.com>

tone_detect can detect an MF tone up to 6 at once. (in practice)


On Mon, Dec 8, 2008 at 11:51 AM, Dan Le <dule.maillist at gmail.com> wrote:

> From my understanding, I didn't think tone_detect detects DTMF since it's
> dual frequencies, rather tone_detect detects single frequencies like fax
> tones.
>
> I would just run an IVR with a session.read or session.getDigits to collect
> DTMF.
>
> Dan
>
> On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com>wrote:
>
>> Same thing with version 10640 build.
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Brian West
>> Sent: Saturday, December 06, 2008 1:01 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] key tone trigger event during call
>>
>> make current or install current svn on a different box.
>>
>> /b
>>
>> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>>
>> >
>> > Ideas?  Am I doing something stupid or is tone_detect not just right
>> > here?
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Mon Dec  8 09:56:30 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 8 Dec 2008 12:56:30 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>
References: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>
	<04ab01c957cf$308e78e0$33014c0a@ws4>
	<914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>
Message-ID: <F8B619FF-4F53-4374-B903-871DB1C938E4@jerris.com>

Are you really trying to detect a tone, or are you trying to detect  
dtmf (could be delivered via rfc2833, info, etc) ?

Mike

On Dec 8, 2008, at 12:51 PM, Dan Le wrote:

> From my understanding, I didn't think tone_detect detects DTMF since  
> it's dual frequencies, rather tone_detect detects single frequencies  
> like fax tones.
>
> I would just run an IVR with a session.read or session.getDigits to  
> collect DTMF.
> Dan
>
>
> On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com>  
> wrote:
> Same thing with version 10640 build.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Saturday, December 06, 2008 1:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
>
> make current or install current svn on a different box.
>
> /b
>
> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>
> >
> > Ideas?  Am I doing something stupid or is tone_detect not just right
> > here?
>

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From john at loopfx.com  Mon Dec  8 13:36:07 2008
From: john at loopfx.com (John Rutherford)
Date: Mon, 8 Dec 2008 16:36:07 -0500
Subject: [Freeswitch-users] No audio after transfer
Message-ID: <81469655CA61444CBB034826ABC6F6E331D5A3@anniesue.loop.local>

I'm trying to get an attended transfer work with freeSWITCH, but it's
not quite working.  I have Microsoft Speech Server on one side and
Televantage on the other.  

 

MSS is originating a call, which freeSWITCH is bridging to Televantage.
That calls connects just fine.  Then, MSS sends a re-INVITE to
Televantage to put the call on hold.  This works too.  Then, MSS
originates another call to freeSWITCH, which is again bridged to
Televantage.  This works fine too.  

 

Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
should be complete, but there is no audio between the two calls-just
silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not
seeing anything obviously wrong.  

 

After the REFER, I can see audio for both calls going between freeSWITCH
and Televantage, so it seems that the only thing missing is freeSWITCH
routing the audio from one call to the other call and vice-versa.

 

 

Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this.  

 

Thanks!

John

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From brian at freeswitch.org  Mon Dec  8 13:48:28 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Dec 2008 15:48:28 -0600
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E331D5A3@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D5A3@anniesue.loop.local>
Message-ID: <244F3A58-EA28-4A76-AD5E-160E65B5B421@freeswitch.org>

Are you on SVN trunk?  If not what rev?

/b

On Dec 8, 2008, at 3:36 PM, John Rutherford wrote:

> Any help would be greatly appreciated.  I have a pcap and the  
> freeSWITCH logs, and I can easily reproduce this.
>
> Thanks!
> John

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From john at loopfx.com  Mon Dec  8 14:16:42 2008
From: john at loopfx.com (John Rutherford)
Date: Mon, 8 Dec 2008 17:16:42 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D5A3@anniesue.loop.local>
	<244F3A58-EA28-4A76-AD5E-160E65B5B421@freeswitch.org>
Message-ID: <81469655CA61444CBB034826ABC6F6E331D5BD@anniesue.loop.local>

Sorry.  I forgot to mention that.  

 

I checked out the trunk last week.  I have revision 10597.  

 

John

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Monday, December 08, 2008 4:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

Are you on SVN trunk?  If not what rev?

 

/b

 

On Dec 8, 2008, at 3:36 PM, John Rutherford wrote:





Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this. 

 

Thanks!

John

 

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From jpalley at idapted.com  Mon Dec  8 20:37:52 2008
From: jpalley at idapted.com (Jonathan Palley)
Date: Tue, 9 Dec 2008 12:37:52 +0800
Subject: [Freeswitch-users] Jitter + Packet Loss
Message-ID: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>

I'm curious to start a discussion on being able to query a channel and get
statistics on the incoming jitter and packet loss (calculated from the RTP,
not RTCP).

Is this on the roadmap?  Is it hard to do?

Would be very useful for us indeed!

Thanks -
JP
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From regs at kinetix.gr  Tue Dec  9 00:21:08 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Tue, 09 Dec 2008 10:21:08 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>
	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
Message-ID: <493E2A74.7010502@kinetix.gr>

"I already added 2 patches for you right.  Just be clear about what you 
want."

And I am grateful of that.

"it is protocol neutral, that's why it starts with sip_"

I didn't know that. I thought that the sip_ variables are protocol 
specific. So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?

"Maybe you should beat around the bush less with your "requirements" for 
your application you are expecting me to support for you."

I am just trying to gather statistics for my providers as I would with 
any VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a 
general necessity for VoIP carriers. It is also very useful for 
troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain 
for, I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot 
the issue better.

"Just be clear about what you want."

I want FS to reach that level of detailing and maturity in all aspects 
so that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by 
the community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The 
patches that you made the last few days *were not intended for
me* exclusively but for *anyone* that will face the same situations 
using FS. If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or 
just use it as an announcement board.

I wish I was a c programmer and get involved with the project actively. 
But I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing 
and suggesting.

Anthony Minessale wrote:
> it is protocol neutral, that's why it starts with sip_
>
> the variable can be any of:
>
> send_bye
> recv_bye
> send_cancel
> send_refuse
>
>
> using that value you can determine the information you asked.  I 
> answered your specific question which was:
> determining "which side hanged up".  Maybe you should beat around the 
> bush less with your "requirements" for your application you are 
> expecting me to support for you.
>
> I already added 2 patches for you right.  Just be clear about what you 
> want.
>
>
>
> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     Not necessarily. For instance I got a "send_cancel" when the
>     calling party hanged up before the other party could pick up.
>     Also, shouldn't something like that be protocol/technology
>     neutral?
>
>
>
>     Anthony Minessale wrote:
>>     sip_hangup_disposition will be set to recv_bye on the side that
>>     was hungup.
>>
>>
>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>>
>>         Any updates about the "which side hanged up" potential variable?
>>
>>         Michael S Collins wrote:
>>>         Makes sense. I will look into this. 
>>>         -MC
>>>
>>>
>>>         On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos
>>>         <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>>>
>>>>         I am sending this second email to request/suggest/enquire
>>>>         about something relevant :
>>>>
>>>>         Wouldn't it be useful to know which end of a specific call
>>>>         leg send the protocol
>>>>         specific hangup cause? Otherwise it would be difficult to
>>>>         understand what really happened.
>>>>
>>>>
>>>>
>>>>         Michael S Collins wrote:
>>>>>         I will do some research on this and let you know what I find out.  
>>>>>         Question: are these internal calls or pstn or ?? Just curious about  
>>>>>         your environment.
>>>>>
>>>>>         Thanks,
>>>>>         MC
>>>>>
>>>>>
>>>>>
>>>>>         On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <mailto:regs at kinetix.gr>regs at kinetix.gr <mailto:regs at kinetix.gr>>  
>>>>>         wrote:
>>>>>
>>>>>           
>>>>>>         The proto_specific_hangup_cause is missing on one of the two
>>>>>>         call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>>         When the callee hangs up it is missing from the b-leg CDR. Is this  
>>>>>>         nornal?
>>>>>>
>>>>>>         And another question : what piece of info could inform me about who
>>>>>>         hanged up?
>>>>>>
>>>>>>
>>>>>>         -- 
>>>>>>         -------------------------------------------
>>>>>>         Apostolos Pantsiopoulos
>>>>>>         Kinetix Tele.com R & D
>>>>>>         email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>>         -------------------------------------------
>>>>>>
>>>>>>
>>>>>>         _______________________________________________
>>>>>>         Freeswitch-users mailing list
>>>>>>         Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>         http://www.freeswitch.org
>>>>>>             
>>>>>         _______________________________________________
>>>>>         Freeswitch-users mailing list
>>>>>         Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>         http://www.freeswitch.org
>>>>>           
>>>>
>>>>
>>>>         -- 
>>>>         -------------------------------------------
>>>>         Apostolos Pantsiopoulos
>>>>         Kinetix Tele.com R & D
>>>>         email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>         ------------------------------------------- 
>>>>         _______________________________________________
>>>>         Freeswitch-users mailing list
>>>>         Freeswitch-users at lists.freeswitch.org
>>>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org
>>>         ------------------------------------------------------------------------
>>>         _______________________________________________
>>>         Freeswitch-users mailing list
>>>         Freeswitch-users at lists.freeswitch.org
>>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         -- 
>>         -------------------------------------------
>>         Apostolos Pantsiopoulos
>>         Kinetix Tele.com R & D
>>         email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>         ------------------------------------------- 
>>
>>
>>         _______________________________________________
>>         Freeswitch-users mailing list
>>         Freeswitch-users at lists.freeswitch.org
>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Anthony Minessale II
>>
>>     FreeSWITCH http://www.freeswitch.org/
>>     ClueCon http://www.cluecon.com/
>>
>>     AIM: anthm
>>     MSN:anthony_minessale at hotmail.com
>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>>     FreeSWITCH Developer Conference
>>     sip:888 at conference.freeswitch.org
>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>     iax:guest at conference.freeswitch.org/888
>>     <http://iax:guest at conference.freeswitch.org/888>
>>     googletalk:conf+888 at conference.freeswitch.org
>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>     pstn:213-799-1400
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From gkuri at ieee.org  Tue Dec  9 00:43:12 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Tue, 09 Dec 2008 00:43:12 -0800
Subject: [Freeswitch-users] root privs for mod_fax
Message-ID: <493E2FA0.5020006@ieee.org>

I've been experimenting with mod_fax and discovered it doesn't appear to
receive faxes unless freeswitch is running as root? it fails trying to
open the tiff file for writing (see the logs below). I'm using the
dialplan as prescribed in the wiki without any changes and the user the
freeswitch process is running under has privs to write to /tmp, but it
still fails to receive faxes. I haven't tried sending any faxes yet.

I'm running r10609.

any ideas?


2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Get document at 14400bps, modem 7
2008-12-09 00:29:41 [WARNING] mod_fax.c:133 spanfax_log_message()
WARNING T.30 Cannot open target TIFF file 'rxfax.tiff'
2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Changing from state 17 to 3
2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Tx:  DCN with final frame tag
2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Tx:  ff 13 fa
2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 HDLC carrier down in state 3
2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D
_TX
2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX
Set rx type 0
2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX
Set tx type 4
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Send complete in phase T30_PHASE_D_TX, state 3
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Send complete in phase T30_PHASE_D_TX, state 3
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Disconnecting
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX
Set rx type 0
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX
Set tx type 1
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Changing from state 3 to 2
2008-12-09 00:29:44 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Send complete in phase T30_PHASE_E, state 2
2008-12-09 00:29:44 [DEBUG] mod_fax.c:163 phase_e_handler()
===============================================================
===============
2008-12-09 00:29:44 [DEBUG] mod_fax.c:176 phase_e_handler() Fax
processing not successful - result (41) TIFF/F file cannot
be opened.



Gabe


From mrjoebain at gmail.com  Tue Dec  9 01:27:46 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 9 Dec 2008 09:27:46 +0000
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
Message-ID: <748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>

On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
> Hi,
>
> I'm writing an IVR in Lua and am having problems dealing with hangups
> cleanly. Very often session:ready() reports true long after I have hung up
> and the hangup hook function I have set doesn't get called either. It
seems
> to report that the session is active indefinitely in some cases where a
loop
> keeps trying to get some dtmf key presses. Is there any trick to using
> session:ready() or the hangup hook that I might have missed?
>
> On a slightly related point I can't seem to access the session properties,
> e.g. session.caller_id_num has a value of nil. Any thoughts here?
>
> Thanks in advance,
>
> Joe Bain
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
2008/12/8 Michael Collins <msc at freeswitch.org>

> Joe,
>
> A few questions... what svn rev are you running? Which operating
> system? Finally, is it possible for you to put your dialplan and Lua
> script up at pastebin.freeswitch.org?
>
> Thanks,
> MC
>
Hi,

I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I can post
the dialplan and lua script though at the moment I can't seem to log in to
the pastebin, I just became a member on the freeswitch homepage but the
pass/username isn't being accepted.

Joe
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From ivan at myrvold.org  Tue Dec  9 01:40:19 2008
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 9 Dec 2008 10:40:19 +0100
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
Message-ID: <1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>

Did you read carefully when asked to provide login and password?
The login and password is there, don't use your own freeswitch login.

Ivan

Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:

> On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
> > Hi,
> >
> > I'm writing an IVR in Lua and am having problems dealing with  
> hangups
> > cleanly. Very often session:ready() reports true long after I have  
> hung up
> > and the hangup hook function I have set doesn't get called either.  
> It seems
> > to report that the session is active indefinitely in some cases  
> where a loop
> > keeps trying to get some dtmf key presses. Is there any trick to  
> using
> > session:ready() or the hangup hook that I might have missed?
> >
> > On a slightly related point I can't seem to access the session  
> properties,
> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
> >
> > Thanks in advance,
> >
> > Joe Bain
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 2008/12/8 Michael Collins <msc at freeswitch.org>
> Joe,
>
> A few questions... what svn rev are you running? Which operating
> system? Finally, is it possible for you to put your dialplan and Lua
> script up at pastebin.freeswitch.org?
>
> Thanks,
> MC
> Hi,
>
> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I  
> can post the dialplan and lua script though at the moment I can't  
> seem to log in to the pastebin, I just became a member on the  
> freeswitch homepage but the pass/username isn't being accepted.
>
> Joe
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mrjoebain at gmail.com  Tue Dec  9 02:06:10 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 9 Dec 2008 10:06:10 +0000
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
	<1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
Message-ID: <748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>

2008/12/9 Ivan C Myrvold <ivan at myrvold.org>

> Did you read carefully when asked to provide login and password? The login
> and password is there, don't use your own freeswitch login.
>
> Ivan
>
>  Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>
>  On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
> > Hi,
> >
> > I'm writing an IVR in Lua and am having problems dealing with hangups
> > cleanly. Very often session:ready() reports true long after I have hung
> up
> > and the hangup hook function I have set doesn't get called either. It
> seems
> > to report that the session is active indefinitely in some cases where a
> loop
> > keeps trying to get some dtmf key presses. Is there any trick to using
> > session:ready() or the hangup hook that I might have missed?
> >
> > On a slightly related point I can't seem to access the session
> properties,
> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
> >
> > Thanks in advance,
> >
> > Joe Bain
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 2008/12/8 Michael Collins <msc at freeswitch.org>
>
>> Joe,
>>
>> A few questions... what svn rev are you running? Which operating
>> system? Finally, is it possible for you to put your dialplan and Lua
>> script up at pastebin.freeswitch.org?
>>
>> Thanks,
>> MC
>>
> Hi,
>
> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I can post
> the dialplan and lua script though at the moment I can't seem to log in to
> the pastebin, I just became a member on the freeswitch homepage but the
> pass/username isn't being accepted.
>
> Joe
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
Ah, I should have read more carefully!

The dialplan is here <http://pastebin.freeswitch.org/6372> and the two
important lua scripts are here <http://pastebin.freeswitch.org/6371> and
here <http://pastebin.freeswitch.org/6370>, the first calls the second. I
didn't include all the Lua script as the problem appears right at the start
(as well as throughout) if the user hangs up when the IVR is speaking
(asking for an id number) then it seems to never get a hangup and loops
trying to get the id number.

Joe
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From helmut.kuper at ewetel.de  Tue Dec  9 02:07:27 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 09 Dec 2008 11:07:27 +0100
Subject: [Freeswitch-users] FS mod_fax
Message-ID: <493E435F.4010402@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I tried to compile mod_fax today with trunk from yesterday. A 'make' in
FS trunk directory led to an error saying that libspandsp.la wasn't
found in libs/spandsp/src. So I had to configure and compile (make)
spandsp manually before compiling FS.

regards
helmut

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.9 (MingW32)

iEYEARECAAYFAkk+Q18ACgkQ4tZeNddg3dw5GgCgmmuLCsAx+T7IzUPayqAXZDaa
BO8AoLa5wUOBqaEG1pOG4Qow8r7J2NF7
=MOUJ
-----END PGP SIGNATURE-----


From helmut.kuper at ewetel.de  Tue Dec  9 02:12:12 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 09 Dec 2008 11:12:12 +0100
Subject: [Freeswitch-users] mod_xml_ldap
Message-ID: <493E447C.5060507@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I tried compile FS with mod_xml_ldap with trunk of yesterday. During
compiling it can't find
http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I looked
there and found that the filename on freeswitch.org side has changed to
http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tar.gz


regards
helmut
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Version: GnuPG v1.4.9 (MingW32)

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iR8AnjypwB2XT/rAYr61yyMXJ+iUY4/d
=iJw/
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From jan.kubr at gmail.com  Tue Dec  9 03:52:24 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Tue, 9 Dec 2008 12:52:24 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
	<5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
	<191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
Message-ID: <698401620812090352x18ef62c2of7abceef9055fb4d@mail.gmail.com>

> btw you can send
>
> call-command: hangup
> hangup-cause: normal_clearing
>
> in place of
> call-command: execute
> execute-app-name: hangup
> execute-app-arg: normal_clearing

What is the difference this makes? Just curious because I've been
using the latter as well.


> we just tested you changes and it works the opposite way it should.
>
> this means: when we do not send an uuid, we get an an error
> (Reply-Text => -ERR invalid session id []). if we send a wrong/not
> existing uuid, the command will be executed on the inbound uuid.

This hasn't been changed, has it? On the latest trunk, if I don't pass
the uuid, I get "-ERR invalid session id". I can always pass it
explicitly though, so no big deal.


Jan Kubr


From jan.kubr at gmail.com  Tue Dec  9 03:53:13 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Tue, 9 Dec 2008 12:53:13 +0100
Subject: [Freeswitch-users] Read app ignores custom variable when called
	via socket interface
In-Reply-To: <191c3a030812080918x6acd7564q3247055cdc941641@mail.gmail.com>
References: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
	<191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>
	<698401620812080248u5f866d7es950d0019686b2c99@mail.gmail.com>
	<191c3a030812080918x6acd7564q3247055cdc941641@mail.gmail.com>
Message-ID: <698401620812090353m75feb135y4327eb42a0f7a4b@mail.gmail.com>

Updated and works great, thanks!

On Mon, Dec 8, 2008 at 6:18 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i added a patch to index the variables on the
> SWITCH_EVENT_CHANNEL_EXECUTE_COMPLETE
> if you want to update
>
> otherwise you can use uuid_getvar to retrieve the variable
>
>
> On Mon, Dec 8, 2008 at 4:48 AM, Jan Kubr <jan.kubr at gmail.com> wrote:
>>
>> OK my bad. The variable is set (I can see it in the Freeswitch console
>> when I use the info app), but they are only not send to me via the
>> socket interface. I get the "variable_*" variables only in the
>> beginning (after calling connect), but not in the events. How do I
>> enable this?
>> Thanks,
>> Jan
>>
>> On Sun, Dec 7, 2008 at 8:44 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > That seems unlikely.
>> > You sure about that?
>> >
>> > The var param is in the middle of the data which is passed as 1 giant
>> > string
>> > to the same exact app execution code.
>> > I don't see how it could differentiate
>> >
>> > did you try executing the info app right after to see all the vars.
>> >
>> > I'm not saying i don't believe you but it seems fishy.
>> >
>> >
>> >
>> > On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr <jan.kubr at gmail.com> wrote:
>> >>
>> >> Hi,
>> >> I checked out the current trunk (rev 10641) and found out that the
>> >> read app ignores the varname parameter, it always puts the result in
>> >> the DTMF-Digit variable. I'm calling it via the socket interface:
>> >>
>> >> sendmsg
>> >> call-command: execute
>> >> execute-app-name: read
>> >> execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res
>> >> 10000 #
>> >> event-lock:true
>> >>
>> >>
>> >> In the XML dialplan it works fine:
>> >>
>> >> <action application="read" data="1 1
>> >> sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
>> >> <action application="phrase" data="spell,${res}"/>
>> >>
>> >>
>> >> I have been using the socket call above successfully in the 1.0.1
>> >> release.
>> >>
>> >> Any ideas? Thanks,
>> >>
>> >> Jan Kubr
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From frank at impactfax.com  Tue Dec  9 05:31:33 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 9 Dec 2008 08:31:33 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <F8B619FF-4F53-4374-B903-871DB1C938E4@jerris.com>
Message-ID: <056d01c95a02$742fe4a0$33014c0a@ws4>

We are actually trying to detect the called party pressing a key - dtmf.
In band for ulaw.  Rfc2833 for 729.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Monday, December 08, 2008 12:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call
 
Are you really trying to detect a tone, or are you trying to detect dtmf
(could be delivered via rfc2833, info, etc) ?
 
Mike
 
On Dec 8, 2008, at 12:51 PM, Dan Le wrote:



>From my understanding, I didn't think tone_detect detects DTMF since
it's dual frequencies, rather tone_detect detects single frequencies
like fax tones. 
I would just run an IVR with a session.read or session.getDigits to
collect DTMF.
Dan
 
On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com>
wrote:
Same thing with version 10640 build.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Saturday, December 06, 2008 1:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call
make current or install current svn on a different box.

/b

On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:

>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?
 
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From frank at impactfax.com  Tue Dec  9 05:38:27 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 9 Dec 2008 08:38:27 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
Message-ID: <057701c95a03$6ac5f610$33014c0a@ws4>

How can FS force a Minimum call duration for a FS caller (someone
calling out of FS)?  
 
We have a carrier that penalizes us with a surcharge for short duration
calls (sound familiar?).
 
So when a FS caller (not a call center or predictive dialer) calls a
cell phone and gets a ring tone or calls an answering machine, the FS
caller hangs up because they do not want to leave a message.  But they
do this in less then a few seconds after the call is answered.  This
becomes a short duration call and bang the surcharge applies.  It is
actually cheaper to pay for a longer call time (6 seconds in this case)
and avoid the short duration surcharge.  But the FS caller does not know
this.
 
So, how can FS hold the connection to the called party open for at least
the minimum amount of time I need to avoid the short call charge. even
though my FS caller has already hung up the phone on his end?  I would
like to do this in the xml dialplan if possible.
 
Thanks
 
-Frank
 
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From jaybinks at gmail.com  Tue Dec  9 05:53:13 2008
From: jaybinks at gmail.com (jay binks)
Date: Tue, 9 Dec 2008 23:53:13 +1000
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
Message-ID: <ec2ce17e0812090553r2f79ae39n4f90cf7dbc4aacd1@mail.gmail.com>

id also love to get any info from the RTCP...
even have this in the XML CDR would be great..
would love to derive quality stats for calls based on RTCP

Jay

On Tue, Dec 9, 2008 at 2:37 PM, Jonathan Palley <jpalley at idapted.com> wrote:

> I'm curious to start a discussion on being able to query a channel and get
> statistics on the incoming jitter and packet loss (calculated from the RTP,
> not RTCP).
>
> Is this on the roadmap?  Is it hard to do?
>
> Would be very useful for us indeed!
>
> Thanks -
> JP
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From brian at freeswitch.org  Tue Dec  9 06:01:59 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 08:01:59 -0600
Subject: [Freeswitch-users] root privs for mod_fax
In-Reply-To: <493E2FA0.5020006@ieee.org>
References: <493E2FA0.5020006@ieee.org>
Message-ID: <7BF70356-892B-45B2-B2D0-B5D3D1B7D01E@freeswitch.org>

If you're running SELinux then you'll need to correct that on your  
machine to allow FreeSWITCH to write to /tmp

/b

On Dec 9, 2008, at 2:43 AM, Gabriel Kuri wrote:

> I've been experimenting with mod_fax and discovered it doesn't  
> appear to
> receive faxes unless freeswitch is running as root? it fails trying to
> open the tiff file for writing (see the logs below). I'm using the
> dialplan as prescribed in the wiki without any changes and the user  
> the
> freeswitch process is running under has privs to write to /tmp, but it
> still fails to receive faxes. I haven't tried sending any faxes yet.
>
> I'm running r10609.
>
> any ideas?



From mrjoebain at gmail.com  Tue Dec  9 06:10:00 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 9 Dec 2008 14:10:00 +0000
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
	<1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
	<748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>
Message-ID: <748d53500812090610t70a11a07u594541a8e132a9d3@mail.gmail.com>

Ok I have been testing more and I have reduced my problem to a pretty short
and simple Lua script. I've posted it at
http://pastebin.freeswitch.org/6373 and
this gets called straight from the dialplan. From my experience so far it
only exits after a caller hangup about 1 in 10 times. Most of the time it
continues to loop until I do 'hupall'.

Thanks in advance if anyone can solve this or offer any advice.

Joe

2008/12/9 Joe Bain <mrjoebain at gmail.com>

> 2008/12/9 Ivan C Myrvold <ivan at myrvold.org>
>
>  Did you read carefully when asked to provide login and password? The
>> login and password is there, don't use your own freeswitch login.
>>
>> Ivan
>>
>>  Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>>
>>  On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
>> > Hi,
>> >
>> > I'm writing an IVR in Lua and am having problems dealing with hangups
>> > cleanly. Very often session:ready() reports true long after I have hung
>> up
>> > and the hangup hook function I have set doesn't get called either. It
>> seems
>> > to report that the session is active indefinitely in some cases where a
>> loop
>> > keeps trying to get some dtmf key presses. Is there any trick to using
>> > session:ready() or the hangup hook that I might have missed?
>> >
>> > On a slightly related point I can't seem to access the session
>> properties,
>> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
>> >
>> > Thanks in advance,
>> >
>> > Joe Bain
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 2008/12/8 Michael Collins <msc at freeswitch.org>
>>
>>> Joe,
>>>
>>> A few questions... what svn rev are you running? Which operating
>>> system? Finally, is it possible for you to put your dialplan and Lua
>>> script up at pastebin.freeswitch.org?
>>>
>>> Thanks,
>>> MC
>>>
>> Hi,
>>
>> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I can post
>> the dialplan and lua script though at the moment I can't seem to log in to
>> the pastebin, I just became a member on the freeswitch homepage but the
>> pass/username isn't being accepted.
>>
>> Joe
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> Ah, I should have read more carefully!
>
> The dialplan is here <http://pastebin.freeswitch.org/6372> and the two
> important lua scripts are here <http://pastebin.freeswitch.org/6371> and
> here <http://pastebin.freeswitch.org/6370>, the first calls the second. I
> didn't include all the Lua script as the problem appears right at the start
> (as well as throughout) if the user hangs up when the IVR is speaking
> (asking for an id number) then it seems to never get a hangup and loops
> trying to get the id number.
>
> Joe
>
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From erick at junctionnetworks.com  Mon Dec  8 17:14:00 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Mon, 08 Dec 2008 20:14:00 -0500
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
Message-ID: <493DC658.8020305@junctionnetworks.com>

Hi There,

I'm trying to get freeswitch to originate all SIP calls through
an outbound proxy.  When I use the originate API command to create a call
to a telephone number I see the SIP packets getting to my proxy just fine.
However if I originate a call to a SIP address then proxy server is 
bypassed,
instead FS is directly messaging the addressee.

Here is the command that I'm trying to use that behaves unexpectedly:
originate sofia/gateway/proxy/alice at bar.com &echo()

However this command produces the results I'm expecting:
originate sofia/gateway/proxy/15551234 &echo()

Here is the result of my sofia status:

freeswitch> sofia status
API CALL [sofia(status)] output:
                     Name          Type                               
Data      State
=================================================================================================
                 external       profile   
sip:mod_sofia at X.X.X.X:5070      RUNNING (0)
                    proxy       gateway   
sip:ejjohnson_ippx at ejjohnson.org      NOREG
=================================================================================================
1 profile 0 aliases

I have also tried setting the sip_invite_domain channel var through  the {}

Could you let me know what I'm doing wrong?

Much appreciated,

Erick J



From scott.ellis at novatex.com.au  Mon Dec  8 21:21:46 2008
From: scott.ellis at novatex.com.au (Scott Ellis)
Date: Tue, 09 Dec 2008 16:21:46 +1100
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
Message-ID: <493E006A.6030507@novatex.com.au>

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From brian at freeswitch.org  Tue Dec  9 06:30:03 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 08:30:03 -0600
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493DC658.8020305@junctionnetworks.com>
References: <493DC658.8020305@junctionnetworks.com>
Message-ID: <26B445AB-3DC6-4A45-B487-2DD7B67B4BA5@freeswitch.org>

First example is WRONG you don't dial via a gateway that way.  If you  
wish to dial alice at bar.com then try sofia/internal/alice at bar.com as  
you don't require a gateway to call alice right?

/b

On Dec 8, 2008, at 7:14 PM, Erick Johnson wrote:

> Here is the command that I'm trying to use that behaves unexpectedly:
> originate sofia/gateway/proxy/alice at bar.com &echo()
>
> However this command produces the results I'm expecting:
> originate sofia/gateway/proxy/15551234 &echo()

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From msc at freeswitch.org  Tue Dec  9 06:44:51 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 06:44:51 -0800
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493E2A74.7010502@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
	<493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
	<493D2B9B.6050205@kinetix.gr>
	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
	<493E2A74.7010502@kinetix.gr>
Message-ID: <87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>

Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.

Do you have FS in production right now? Just curious.

Thanks,
MC

On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs at kinetix.gr> wrote:
> "I already added 2 patches for you right.  Just be clear about what you
> want."
>
> And I am grateful of that.
>
> "it is protocol neutral, that's why it starts with sip_"
>
> I didn't know that. I thought that the sip_ variables are protocol specific.
> So one would expect there to be an iax_hangup_disposition,
> woomera_hangup_disposition etc?
>
> "Maybe you should beat around the bush less with your "requirements" for
> your application you are expecting me to support for you."
>
> I am just trying to gather statistics for my providers as I would with any
> VoIP softswitch. (hangup causes per terminator per destination)
> I don't think that this is a specific "application" rather than a general
> necessity for VoIP carriers. It is also very useful for troubleshooting
> purposes : when I look at my CDRs to find a call that I got a complain for,
> I want to be able to tell if it was me or the provider who
> hanged up and gave a specific hangup cause, so that I can troubleshoot the
> issue better.
>
> "Just be clear about what you want."
>
> I want FS to reach that level of detailing and maturity in all aspects so
> that it could be the softswitch of choice by any VoIP entrepreneur
> (or hobbyist) and it is my strong belief that this can only be done by the
> community giving feedback to the programmers about what
> they find useful or not (i.e. experience from real-life situations). The
> patches that you made the last few days were not intended for
> me exclusively but for anyone that will face the same situations using FS.
> If you want the community to stop sending feedback about
> features/improvements you may as well close down this mailing list or just
> use it as an announcement board.
>
> I wish I was a c programmer and get involved with the project actively. But
> I am not. And as far as I can tell most of the registered users
> in this list aren't either. So they only way we can help is by testing and
> suggesting.
>
> Anthony Minessale wrote:
>
> it is protocol neutral, that's why it starts with sip_
>
> the variable can be any of:
>
> send_bye
> recv_bye
> send_cancel
> send_refuse
>
>
> using that value you can determine the information you asked.  I answered
> your specific question which was:
> determining "which side hanged up".  Maybe you should beat around the bush
> less with your "requirements" for your application you are expecting me to
> support for you.
>
> I already added 2 patches for you right.  Just be clear about what you want.
>
>
>
> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
> wrote:
>>
>> Not necessarily. For instance I got a "send_cancel" when the
>> calling party hanged up before the other party could pick up.
>> Also, shouldn't something like that be protocol/technology
>> neutral?
>>
>>
>>
>> Anthony Minessale wrote:
>>
>> sip_hangup_disposition will be set to recv_bye on the side that was
>> hungup.
>>
>>
>> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>> wrote:
>>>
>>> Any updates about the "which side hanged up" potential variable?
>>>
>>> Michael S Collins wrote:
>>>
>>> Makes sense. I will look into this.
>>> -MC
>>>
>>>
>>> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>> wrote:
>>>
>>> I am sending this second email to request/suggest/enquire about something
>>> relevant :
>>>
>>> Wouldn't it be useful to know which end of a specific call leg send the
>>> protocol
>>> specific hangup cause? Otherwise it would be difficult to understand what
>>> really happened.
>>>
>>>
>>>
>>> Michael S Collins wrote:
>>>
>>> I will do some research on this and let you know what I find out.
>>> Question: are these internal calls or pstn or ?? Just curious about
>>> your environment.
>>>
>>> Thanks,
>>> MC
>>>
>>>
>>>
>>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>> wrote:
>>>
>>>
>>>
>>> The proto_specific_hangup_cause is missing on one of the two
>>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>>> When the callee hangs up it is missing from the b-leg CDR. Is this
>>> nornal?
>>>
>>> And another question : what piece of info could inform me about who
>>> hanged up?
>>>
>>>
>>> --
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> ________________________________
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> ________________________________
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> ________________________________
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Tue Dec  9 06:50:07 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 06:50:07 -0800
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493E435F.4010402@ewetel.de>
References: <493E435F.4010402@ewetel.de>
Message-ID: <87f2f3b90812090650g67bd17a2w5a1e7490c1617abf@mail.gmail.com>

Which OS are you running?
-MC

On Tue, Dec 9, 2008 at 2:07 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I tried to compile mod_fax today with trunk from yesterday. A 'make' in
> FS trunk directory led to an error saying that libspandsp.la wasn't
> found in libs/spandsp/src. So I had to configure and compile (make)
> spandsp manually before compiling FS.
>
> regards
> helmut
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAkk+Q18ACgkQ4tZeNddg3dw5GgCgmmuLCsAx+T7IzUPayqAXZDaa
> BO8AoLa5wUOBqaEG1pOG4Qow8r7J2NF7
> =MOUJ
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Tue Dec  9 06:51:08 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 06:51:08 -0800
Subject: [Freeswitch-users] mod_xml_ldap
In-Reply-To: <493E447C.5060507@ewetel.de>
References: <493E447C.5060507@ewetel.de>
Message-ID: <87f2f3b90812090651p671d7a0fgd3f1bafd4f418f40@mail.gmail.com>

Thanks again for the heads up. We'll check it out.
-MC

On Tue, Dec 9, 2008 at 2:12 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I tried compile FS with mod_xml_ldap with trunk of yesterday. During
> compiling it can't find
> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I looked
> there and found that the filename on freeswitch.org side has changed to
> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tar.gz
>
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAkk+RHwACgkQ4tZeNddg3dyp7wCeN7fvIj4OSH1rsuglD46qtS36
> iR8AnjypwB2XT/rAYr61yyMXJ+iUY4/d
> =iJw/
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gilbertandrew at me.com  Tue Dec  9 07:08:09 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 09 Dec 2008 10:08:09 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <057701c95a03$6ac5f610$33014c0a@ws4>
References: <057701c95a03$6ac5f610$33014c0a@ws4>
Message-ID: <4CF7C22E-64D4-4EEF-8153-4999E06E3A9F@me.com>

Don't want the tone to be wrong here, but this makes no sense.  
Carriers surcharge like this precisely to guard against call center,  
predictive and other mass outbound calling scenarios.

It just doesn't make since, math wise, that individuals hanging up on  
voice mail are going to significantly impact overall ACD stats, etc.  
So unless you have a very strange set of use cases or are pushing  
another category of traffic (ie call center) that skews you overall  
relationship with the carrier - I would go back and re-negotiate your  
arrangement.

Yes, FS is a b2bua and all is possible. But it is probably a better  
use of time to approach this as a business issue.

My 2 cents.


On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:

> How can FS force a Minimum call duration for a FS caller (someone  
> calling out of FS)?
>
> We have a carrier that penalizes us with a surcharge for short  
> duration calls (sound familiar?).
>
> So when a FS caller (not a call center or predictive dialer) calls a  
> cell phone and gets a ring tone or calls an answering machine, the  
> FS caller hangs up because they do not want to leave a message.  But  
> they do this in less then a few seconds after the call is answered.   
> This becomes a short duration call and bang the surcharge applies.   
> It is actually cheaper to pay for a longer call time (6 seconds in  
> this case) and avoid the short duration surcharge.  But the FS  
> caller does not know this.
>
> So, how can FS hold the connection to the called party open for at  
> least the minimum amount of time I need to avoid the short call  
> charge? even though my FS caller has already hung up the phone on  
> his end?  I would like to do this in the xml dialplanif possible.
>
> Thanks
>
> -Frank
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From regs at kinetix.gr  Tue Dec  9 07:19:26 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Tue, 09 Dec 2008 17:19:26 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>
	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>
Message-ID: <493E8C7E.1050306@kinetix.gr>


    We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the  CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.

    We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last few 
years
with various platforms and troubleshooting experiences with other VoIP 
carriers.



Michael Collins wrote:
> Thanks for your feedback. It definitely helps to know not only what
> you need FS to do but why you need it to do so.
>
> Do you have FS in production right now? Just curious.
>
> Thanks,
> MC
>
> On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
> <regs at kinetix.gr> wrote:
>   
>> "I already added 2 patches for you right.  Just be clear about what you
>> want."
>>
>> And I am grateful of that.
>>
>> "it is protocol neutral, that's why it starts with sip_"
>>
>> I didn't know that. I thought that the sip_ variables are protocol specific.
>> So one would expect there to be an iax_hangup_disposition,
>> woomera_hangup_disposition etc?
>>
>> "Maybe you should beat around the bush less with your "requirements" for
>> your application you are expecting me to support for you."
>>
>> I am just trying to gather statistics for my providers as I would with any
>> VoIP softswitch. (hangup causes per terminator per destination)
>> I don't think that this is a specific "application" rather than a general
>> necessity for VoIP carriers. It is also very useful for troubleshooting
>> purposes : when I look at my CDRs to find a call that I got a complain for,
>> I want to be able to tell if it was me or the provider who
>> hanged up and gave a specific hangup cause, so that I can troubleshoot the
>> issue better.
>>
>> "Just be clear about what you want."
>>
>> I want FS to reach that level of detailing and maturity in all aspects so
>> that it could be the softswitch of choice by any VoIP entrepreneur
>> (or hobbyist) and it is my strong belief that this can only be done by the
>> community giving feedback to the programmers about what
>> they find useful or not (i.e. experience from real-life situations). The
>> patches that you made the last few days were not intended for
>> me exclusively but for anyone that will face the same situations using FS.
>> If you want the community to stop sending feedback about
>> features/improvements you may as well close down this mailing list or just
>> use it as an announcement board.
>>
>> I wish I was a c programmer and get involved with the project actively. But
>> I am not. And as far as I can tell most of the registered users
>> in this list aren't either. So they only way we can help is by testing and
>> suggesting.
>>
>> Anthony Minessale wrote:
>>
>> it is protocol neutral, that's why it starts with sip_
>>
>> the variable can be any of:
>>
>> send_bye
>> recv_bye
>> send_cancel
>> send_refuse
>>
>>
>> using that value you can determine the information you asked.  I answered
>> your specific question which was:
>> determining "which side hanged up".  Maybe you should beat around the bush
>> less with your "requirements" for your application you are expecting me to
>> support for you.
>>
>> I already added 2 patches for you right.  Just be clear about what you want.
>>
>>
>>
>> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>> wrote:
>>     
>>> Not necessarily. For instance I got a "send_cancel" when the
>>> calling party hanged up before the other party could pick up.
>>> Also, shouldn't something like that be protocol/technology
>>> neutral?
>>>
>>>
>>>
>>> Anthony Minessale wrote:
>>>
>>> sip_hangup_disposition will be set to recv_bye on the side that was
>>> hungup.
>>>
>>>
>>> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>> wrote:
>>>       
>>>> Any updates about the "which side hanged up" potential variable?
>>>>
>>>> Michael S Collins wrote:
>>>>
>>>> Makes sense. I will look into this.
>>>> -MC
>>>>
>>>>
>>>> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>>> wrote:
>>>>
>>>> I am sending this second email to request/suggest/enquire about something
>>>> relevant :
>>>>
>>>> Wouldn't it be useful to know which end of a specific call leg send the
>>>> protocol
>>>> specific hangup cause? Otherwise it would be difficult to understand what
>>>> really happened.
>>>>
>>>>
>>>>
>>>> Michael S Collins wrote:
>>>>
>>>> I will do some research on this and let you know what I find out.
>>>> Question: are these internal calls or pstn or ?? Just curious about
>>>> your environment.
>>>>
>>>> Thanks,
>>>> MC
>>>>
>>>>
>>>>
>>>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>>> wrote:
>>>>
>>>>
>>>>
>>>> The proto_specific_hangup_cause is missing on one of the two
>>>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>> When the callee hangs up it is missing from the b-leg CDR. Is this
>>>> nornal?
>>>>
>>>> And another question : what piece of info could inform me about who
>>>> hanged up?
>>>>
>>>>
>>>> --
>>>> -------------------------------------------
>>>> Apostolos Pantsiopoulos
>>>> Kinetix Tele.com R & D
>>>> email: regs at kinetix.gr
>>>> -------------------------------------------
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> -------------------------------------------
>>>> Apostolos Pantsiopoulos
>>>> Kinetix Tele.com R & D
>>>> email: regs at kinetix.gr
>>>> -------------------------------------------
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> ________________________________
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> -------------------------------------------
>>>> Apostolos Pantsiopoulos
>>>> Kinetix Tele.com R & D
>>>> email: regs at kinetix.gr
>>>> -------------------------------------------
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>         
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:213-799-1400
>>>
>>> ________________________________
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> ________________________________
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From mike at jerris.com  Tue Dec  9 07:45:56 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 10:45:56 -0500
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
Message-ID: <B7BC254D-855D-4247-82B9-BEF52C024E9E@jerris.com>

It is something we have been discussing as we need these stats to do  
rtcp properly but we have not written any code to do so.  It is  
"somewhat" difficult.  I would say it is on our minds but not on any  
roadmap just yet.

MIke

On Dec 8, 2008, at 11:37 PM, Jonathan Palley wrote:

> I'm curious to start a discussion on being able to query a channel  
> and get statistics on the incoming jitter and packet loss  
> (calculated from the RTP, not RTCP).
>
> Is this on the roadmap?  Is it hard to do?



From mike at jerris.com  Tue Dec  9 07:56:32 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 10:56:32 -0500
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493E435F.4010402@ewetel.de>
References: <493E435F.4010402@ewetel.de>
Message-ID: <FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>

make sure you have libtiff and libtiff dev packages installed then re- 
configure freeswitch

Mike

On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I tried to compile mod_fax today with trunk from yesterday. A 'make'  
> in
> FS trunk directory led to an error saying that libspandsp.la wasn't
> found in libs/spandsp/src. So I had to configure and compile (make)
> spandsp manually before compiling FS.
>
> regards
> helmut
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAkk+Q18ACgkQ4tZeNddg3dw5GgCgmmuLCsAx+T7IzUPayqAXZDaa
> BO8AoLa5wUOBqaEG1pOG4Qow8r7J2NF7
> =MOUJ
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Tue Dec  9 07:59:50 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 10:59:50 -0500
Subject: [Freeswitch-users] mod_xml_ldap
In-Reply-To: <87f2f3b90812090651p671d7a0fgd3f1bafd4f418f40@mail.gmail.com>
References: <493E447C.5060507@ewetel.de>
	<87f2f3b90812090651p671d7a0fgd3f1bafd4f418f40@mail.gmail.com>
Message-ID: <89D6643E-06C5-45B2-8244-D211C857B182@jerris.com>

Fixed in svn r10678.  Thanks for the report.

Mike

On Dec 9, 2008, at 9:51 AM, Michael Collins wrote:

> Thanks again for the heads up. We'll check it out.
> -MC
>
> On Tue, Dec 9, 2008 at 2:12 AM, Helmut Kuper  
> <helmut.kuper at ewetel.de> wrote:
>> I tried compile FS with mod_xml_ldap with trunk of yesterday. During
>> compiling it can't find
>> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I  
>> looked
>> there and found that the filename on freeswitch.org side has  
>> changed to
>> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tar.gz


From mike at jerris.com  Tue Dec  9 08:01:51 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 11:01:51 -0500
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <748d53500812090610t70a11a07u594541a8e132a9d3@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
	<1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
	<748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>
	<748d53500812090610t70a11a07u594541a8e132a9d3@mail.gmail.com>
Message-ID: <8F96510A-1A1E-45E9-A206-FDA66CAEA06F@jerris.com>


On Dec 9, 2008, at 9:10 AM, Joe Bain wrote:

> Ok I have been testing more and I have reduced my problem to a  
> pretty short and simple Lua script. I've posted it at http://pastebin.freeswitch.org/6373 
>  and this gets called straight from the dialplan. From my experience  
> so far it only exits after a caller hangup about 1 in 10 times. Most  
> of the time it continues to loop until I do 'hupall'.
>
> Thanks in advance if anyone can solve this or offer any advice.
>
> Joe
>
> 2008/12/9 Joe Bain <mrjoebain at gmail.com>
> 2008/12/9 Ivan C Myrvold <ivan at myrvold.org>
>
> Did you read carefully when asked to provide login and password?
> The login and password is there, don't use your own freeswitch login.
>
> Ivan
>
> Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>
>> On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
>> > Hi,
>> >
>> > I'm writing an IVR in Lua and am having problems dealing with  
>> hangups
>> > cleanly. Very often session:ready() reports true long after I  
>> have hung up
>> > and the hangup hook function I have set doesn't get called  
>> either. It seems
>> > to report that the session is active indefinitely in some cases  
>> where a loop
>> > keeps trying to get some dtmf key presses. Is there any trick to  
>> using
>> > session:ready() or the hangup hook that I might have missed?
>> >
>> > On a slightly related point I can't seem to access the session  
>> properties,
>> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
>> >

>> Joe,
>>
>> A few questions... what svn rev are you running? Which operating
>> system? Finally, is it possible for you to put your dialplan and Lua
>> script up at pastebin.freeswitch.org?
>>
>> Thanks,
>> MC
>> Hi,
>>
>> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I  
>> can post the dialplan and lua script though at the moment I can't  
>> seem to log in to the pastebin, I just became a member on the  
>> freeswitch homepage but the pass/username isn't being accepted.
>>
>> Joe
>
>
> Ah, I should have read more carefully!
>
> The dialplan is here and the two important lua scripts are here and  
> here, the first calls the second. I didn't include all the Lua  
> script as the problem appears right at the start (as well as  
> throughout) if the user hangs up when the IVR is speaking (asking  
> for an id number) then it seems to never get a hangup and loops  
> trying to get the id number.
>
> Joe

We just tested this with current svn trunk and it appears to work  
fine, could you try updating and see if it is still a problem for you

Mike

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From mike at jerris.com  Tue Dec  9 08:06:02 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 11:06:02 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <056d01c95a02$742fe4a0$33014c0a@ws4>
References: <056d01c95a02$742fe4a0$33014c0a@ws4>
Message-ID: <4B77D066-6FA0-4689-B9A7-B47B4429403D@jerris.com>

you don't want to be using the tone detect here, you want to be using  
bind_meta, but without the meta key, which I don't think it can  
actually do currently.

Mike

On Dec 9, 2008, at 8:31 AM, Frank @ Impact wrote:

> We are actually trying to detect the called party pressing a key ?  
> dtmf.  In band for ulaw.  Rfc2833 for 729.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Jerris
> Sent: Monday, December 08, 2008 12:57 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
>
> Are you really trying to detect a tone, or are you trying to detect  
> dtmf (could be delivered via rfc2833, info, etc) ?
>
> Mike
>
> On Dec 8, 2008, at 12:51 PM, Dan Le wrote:
>
>
> From my understanding, I didn't think tone_detect detects DTMF since  
> it's dual frequencies, rather tone_detect detects single frequencies  
> like fax tones.
>
> I would just run an IVR with a session.read or session.getDigits to  
> collect DTMF.
>
> Dan
>
>
> On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com>  
> wrote:
> Same thing with version 10640 build.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Saturday, December 06, 2008 1:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
> make current or install current svn on a different box.
>
> /b
>
> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>
> >
> > Ideas?  Am I doing something stupid or is tone_detect not just right
> > here?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From helmut.kuper at ewetel.de  Tue Dec  9 08:09:10 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 09 Dec 2008 17:09:10 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>
References: <493E435F.4010402@ewetel.de>
	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>
Message-ID: <493E9826.108@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

don't know if you get me right: Everything is there, but obviously FS
makefile has to compile "libs/spandsp/src" before mod_fax (at least I
guess so). Currently the Makefile referred to libspandsp.la before it is
compiled.

regards
helmut



Michael Jerris schrieb:
> make sure you have libtiff and libtiff dev packages installed then re- 
> configure freeswitch
> 
> Mike
> 
> On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:
> 
> Hello,
> 
> I tried to compile mod_fax today with trunk from yesterday. A 'make'  
> in
> FS trunk directory led to an error saying that libspandsp.la wasn't
> found in libs/spandsp/src. So I had to configure and compile (make)
> spandsp manually before compiling FS.
> 
> regards
> helmut
> 
>>
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From mike at jerris.com  Tue Dec  9 08:10:06 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 11:10:06 -0500
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <698401620812090352x18ef62c2of7abceef9055fb4d@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
	<5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
	<191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
	<698401620812090352x18ef62c2of7abceef9055fb4d@mail.gmail.com>
Message-ID: <4CBD44BD-3824-4BFD-BAC1-6E1DAE0C71E7@jerris.com>


On Dec 9, 2008, at 6:52 AM, Jan Kubr wrote:

>> btw you can send
>>
>> call-command: hangup
>> hangup-cause: normal_clearing
>>
>> in place of
>> call-command: execute
>> execute-app-name: hangup
>> execute-app-arg: normal_clearing
>
> What is the difference this makes? Just curious because I've been
> using the latter as well.
>
>
>> we just tested you changes and it works the opposite way it should.
>>
>> this means: when we do not send an uuid, we get an an error
>> (Reply-Text => -ERR invalid session id []). if we send a wrong/not
>> existing uuid, the command will be executed on the inbound uuid.
>
> This hasn't been changed, has it? On the latest trunk, if I don't pass
> the uuid, I get "-ERR invalid session id". I can always pass it
> explicitly though, so no big deal.
>
>
> Jan Kubr

We did have confirmation from others that this is working properly  
now.  Can you please make sure you are on current trunk and re-test  
this.

Mike



From sicfslist at gmail.com  Tue Dec  9 08:12:01 2008
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Tue, 9 Dec 2008 10:12:01 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493E8C7E.1050306@kinetix.gr>
References: <49391D40.6050103@kinetix.gr> <49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
	<493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
	<493D2B9B.6050205@kinetix.gr>
	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
	<493E2A74.7010502@kinetix.gr>
	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>
	<493E8C7E.1050306@kinetix.gr>
Message-ID: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>

Hello,
This is just my 2 cents ... but my experience has been that trying to catch
all of the various variables (i.e. from XML_CDR) or otherwise can be a
little trying (a row in your CDR database could be over 100 fields long!).

The best option here is to catch the UUID's for the 2 call legs, capture all
SIP messaging, parse and dump the messaging, and then correlate the calls
from the CDR from there.

Much easier than trying to do it from FS ... and most folks want to see SIP
captures anyway (very broad set of tools to debug).

Measuring things like ASR, PDD, etc in my opinion is much easier from the
raw messaging than trying to do something with FS CDR records.



On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>
>     We are currently in the migration process from our
> current system to a FS based setup. We are in the process of
> adapting our billing and routing to FS. All the  CDRs (and variables)
> related issues that we have been discussing on this mailing list
> come from the need to extract the same level of information from FS as
> we do with our current closed source proprietary system. So, we
> chose FS because of the versatility it provides in every aspect (event
> handling, config implementation etc.) and we strongly believe that all
> these additions/fixes would be beneficial to many potential FS users.
>
>     We are at your disposal for more details in case you need
> more information about what exactly we are trying to do. Basically,
> our approach is from the VoIP carrier's point of view rather than the
> PBX user's/implementor's. So, the details that we asked to be introduced
> to FS come from real life issues that we have faced during the last few
> years
> with various platforms and troubleshooting experiences with other VoIP
> carriers.
>
>
>
>
> Michael Collins wrote:
>
> Thanks for your feedback. It definitely helps to know not only what
> you need FS to do but why you need it to do so.
>
> Do you have FS in production right now? Just curious.
>
> Thanks,
> MC
>
> On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos<regs at kinetix.gr> <regs at kinetix.gr> wrote:
>
>
>  "I already added 2 patches for you right.  Just be clear about what you
> want."
>
> And I am grateful of that.
>
> "it is protocol neutral, that's why it starts with sip_"
>
> I didn't know that. I thought that the sip_ variables are protocol specific.
> So one would expect there to be an iax_hangup_disposition,
> woomera_hangup_disposition etc?
>
> "Maybe you should beat around the bush less with your "requirements" for
> your application you are expecting me to support for you."
>
> I am just trying to gather statistics for my providers as I would with any
> VoIP softswitch. (hangup causes per terminator per destination)
> I don't think that this is a specific "application" rather than a general
> necessity for VoIP carriers. It is also very useful for troubleshooting
> purposes : when I look at my CDRs to find a call that I got a complain for,
> I want to be able to tell if it was me or the provider who
> hanged up and gave a specific hangup cause, so that I can troubleshoot the
> issue better.
>
> "Just be clear about what you want."
>
> I want FS to reach that level of detailing and maturity in all aspects so
> that it could be the softswitch of choice by any VoIP entrepreneur
> (or hobbyist) and it is my strong belief that this can only be done by the
> community giving feedback to the programmers about what
> they find useful or not (i.e. experience from real-life situations). The
> patches that you made the last few days were not intended for
> me exclusively but for anyone that will face the same situations using FS.
> If you want the community to stop sending feedback about
> features/improvements you may as well close down this mailing list or just
> use it as an announcement board.
>
> I wish I was a c programmer and get involved with the project actively. But
> I am not. And as far as I can tell most of the registered users
> in this list aren't either. So they only way we can help is by testing and
> suggesting.
>
> Anthony Minessale wrote:
>
> it is protocol neutral, that's why it starts with sip_
>
> the variable can be any of:
>
> send_bye
> recv_bye
> send_cancel
> send_refuse
>
>
> using that value you can determine the information you asked.  I answered
> your specific question which was:
> determining "which side hanged up".  Maybe you should beat around the bush
> less with your "requirements" for your application you are expecting me to
> support for you.
>
> I already added 2 patches for you right.  Just be clear about what you want.
>
>
>
> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <regs at kinetix.gr>
> wrote:
>
>
>  Not necessarily. For instance I got a "send_cancel" when the
> calling party hanged up before the other party could pick up.
> Also, shouldn't something like that be protocol/technology
> neutral?
>
>
>
> Anthony Minessale wrote:
>
> sip_hangup_disposition will be set to recv_bye on the side that was
> hungup.
>
>
> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <regs at kinetix.gr>
> wrote:
>
>
>  Any updates about the "which side hanged up" potential variable?
>
> Michael S Collins wrote:
>
> Makes sense. I will look into this.
> -MC
>
>
> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <regs at kinetix.gr>
> wrote:
>
> I am sending this second email to request/suggest/enquire about something
> relevant :
>
> Wouldn't it be useful to know which end of a specific call leg send the
> protocol
> specific hangup cause? Otherwise it would be difficult to understand what
> really happened.
>
>
>
> Michael S Collins wrote:
>
> I will do some research on this and let you know what I find out.
> Question: are these internal calls or pstn or ?? Just curious about
> your environment.
>
> Thanks,
> MC
>
>
>
> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <regs at kinetix.gr>
> wrote:
>
>
>
> The proto_specific_hangup_cause is missing on one of the two
> call legs. When the caller hangs up it is missing from the a-leg CDR.
> When the callee hangs up it is missing from the b-leg CDR. Is this
> nornal?
>
> And another question : what piece of info could inform me about who
> hanged up?
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
> ________________________________
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>          --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthmMSN:anthony_minessale at hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conferencesip:888 at conference.freeswitch.orgiax:guest at conference.freeswitch.org/888googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> ________________________________
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>        --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthmMSN:anthony_minessale at hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conferencesip:888 at conference.freeswitch.orgiax:guest at conference.freeswitch.org/888googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> ________________________________
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>      _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Dec  9 08:25:22 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 11:25:22 -0500
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493E9826.108@ewetel.de>
References: <493E435F.4010402@ewetel.de>
	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>
	<493E9826.108@ewetel.de>
Message-ID: <724B6B1A-4834-4907-A9E8-73076981176C@jerris.com>


On Dec 9, 2008, at 11:09 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> don't know if you get me right: Everything is there, but obviously FS
> makefile has to compile "libs/spandsp/src" before mod_fax (at least I
> guess so). Currently the Makefile referred to libspandsp.la before  
> it is
> compiled.
>
> regards
> helmut

Did you try it?

Mike



From gkuri at ieee.org  Tue Dec  9 08:35:25 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Tue, 09 Dec 2008 08:35:25 -0800
Subject: [Freeswitch-users] root privs for mod_fax
In-Reply-To: <7BF70356-892B-45B2-B2D0-B5D3D1B7D01E@freeswitch.org>
References: <493E2FA0.5020006@ieee.org>
	<7BF70356-892B-45B2-B2D0-B5D3D1B7D01E@freeswitch.org>
Message-ID: <493E9E4D.7040809@ieee.org>

I'm running Gentoo Linux.

# uname -a

Linux 2.6.25-gentoo-r7 #1 SMP PREEMPT Sun Oct 5 01:51:24 PDT 2008 x86_64
Intel(R) Xeon(R) CPU X3320 @ 2.50GHz GenuineIntel GNU/Linux

/tmp is writable by everyone ...

# ls -ld /tmp
drwxrwxrwt 4 root root 4096 Dec  9 08:28 /tmp

ideas?

also, I assume the spool directory is just where it's spooling the file
temporarily while the fax is coming in, but is there a variable I can
set to tell it where to put the file after the fax has been received? it
looks like the time I ran FS as root and received a fax successfully,
the tiff file ended up in /root ?

Gabe

Brian West wrote:
> If you're running SELinux then you'll need to correct that on your  
> machine to allow FreeSWITCH to write to /tmp
> 
> /b
> 
> On Dec 9, 2008, at 2:43 AM, Gabriel Kuri wrote:
> 
>> I've been experimenting with mod_fax and discovered it doesn't  
>> appear to
>> receive faxes unless freeswitch is running as root? it fails trying to
>> open the tiff file for writing (see the logs below). I'm using the
>> dialplan as prescribed in the wiki without any changes and the user  
>> the
>> freeswitch process is running under has privs to write to /tmp, but it
>> still fails to receive faxes. I haven't tried sending any faxes yet.
>>
>> I'm running r10609.
>>
>> any ideas?
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 





From regs at kinetix.gr  Tue Dec  9 08:37:25 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Tue, 09 Dec 2008 18:37:25 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>
	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
Message-ID: <493E9EC5.1060701@kinetix.gr>

That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).

In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.



Shelby Ramsey wrote:
> Hello,
>
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise 
> can be a little trying (a row in your CDR database could be over 100 
> fields long!).  
>
> The best option here is to catch the UUID's for the 2 call legs, 
> capture all SIP messaging, parse and dump the messaging, and then 
> correlate the calls from the CDR from there.  
>
> Much easier than trying to do it from FS ... and most folks want to 
> see SIP captures anyway (very broad set of tools to debug).  
>
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.
>
>
>
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
>
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be
>     introduced
>     to FS come from real life issues that we have faced during the
>     last few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
>
>
>
>
>     Michael Collins wrote:
>>     Thanks for your feedback. It definitely helps to know not only what
>>     you need FS to do but why you need it to do so.
>>
>>     Do you have FS in production right now? Just curious.
>>
>>     Thanks,
>>     MC
>>
>>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "I already added 2 patches for you right.  Just be clear about what you
>>>     want."
>>>
>>>     And I am grateful of that.
>>>
>>>     "it is protocol neutral, that's why it starts with sip_"
>>>
>>>     I didn't know that. I thought that the sip_ variables are protocol specific.
>>>     So one would expect there to be an iax_hangup_disposition,
>>>     woomera_hangup_disposition etc?
>>>
>>>     "Maybe you should beat around the bush less with your "requirements" for
>>>     your application you are expecting me to support for you."
>>>
>>>     I am just trying to gather statistics for my providers as I would with any
>>>     VoIP softswitch. (hangup causes per terminator per destination)
>>>     I don't think that this is a specific "application" rather than a general
>>>     necessity for VoIP carriers. It is also very useful for troubleshooting
>>>     purposes : when I look at my CDRs to find a call that I got a complain for,
>>>     I want to be able to tell if it was me or the provider who
>>>     hanged up and gave a specific hangup cause, so that I can troubleshoot the
>>>     issue better.
>>>
>>>     "Just be clear about what you want."
>>>
>>>     I want FS to reach that level of detailing and maturity in all aspects so
>>>     that it could be the softswitch of choice by any VoIP entrepreneur
>>>     (or hobbyist) and it is my strong belief that this can only be done by the
>>>     community giving feedback to the programmers about what
>>>     they find useful or not (i.e. experience from real-life situations). The
>>>     patches that you made the last few days were not intended for
>>>     me exclusively but for anyone that will face the same situations using FS.
>>>     If you want the community to stop sending feedback about
>>>     features/improvements you may as well close down this mailing list or just
>>>     use it as an announcement board.
>>>
>>>     I wish I was a c programmer and get involved with the project actively. But
>>>     I am not. And as far as I can tell most of the registered users
>>>     in this list aren't either. So they only way we can help is by testing and
>>>     suggesting.
>>>
>>>     Anthony Minessale wrote:
>>>
>>>     it is protocol neutral, that's why it starts with sip_
>>>
>>>     the variable can be any of:
>>>
>>>     send_bye
>>>     recv_bye
>>>     send_cancel
>>>     send_refuse
>>>
>>>
>>>     using that value you can determine the information you asked.  I answered
>>>     your specific question which was:
>>>     determining "which side hanged up".  Maybe you should beat around the bush
>>>     less with your "requirements" for your application you are expecting me to
>>>     support for you.
>>>
>>>     I already added 2 patches for you right.  Just be clear about what you want.
>>>
>>>
>>>
>>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     Not necessarily. For instance I got a "send_cancel" when the
>>>>     calling party hanged up before the other party could pick up.
>>>>     Also, shouldn't something like that be protocol/technology
>>>>     neutral?
>>>>
>>>>
>>>>
>>>>     Anthony Minessale wrote:
>>>>
>>>>     sip_hangup_disposition will be set to recv_bye on the side that was
>>>>     hungup.
>>>>
>>>>
>>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     Any updates about the "which side hanged up" potential variable?
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     Makes sense. I will look into this.
>>>>>     -MC
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>     I am sending this second email to request/suggest/enquire about something
>>>>>     relevant :
>>>>>
>>>>>     Wouldn't it be useful to know which end of a specific call leg send the
>>>>>     protocol
>>>>>     specific hangup cause? Otherwise it would be difficult to understand what
>>>>>     really happened.
>>>>>
>>>>>
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     I will do some research on this and let you know what I find out.
>>>>>     Question: are these internal calls or pstn or ?? Just curious about
>>>>>     your environment.
>>>>>
>>>>>     Thanks,
>>>>>     MC
>>>>>
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>
>>>>>
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>     ________________________________
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>         
>>     _______________________________________________
>>     Freeswitch-users mailing list
>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From msc at freeswitch.org  Tue Dec  9 08:37:55 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 08:37:55 -0800
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493E9826.108@ewetel.de>
References: <493E435F.4010402@ewetel.de>
	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>
	<493E9826.108@ewetel.de>
Message-ID: <87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>

Helmut,
I think Mike J was pointing out that spandsp needs libtiff and
libtiff-devel in order to compile, so you need to do that first and
then compile freeswitch.
-MC

On Tue, Dec 9, 2008 at 8:09 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> don't know if you get me right: Everything is there, but obviously FS
> makefile has to compile "libs/spandsp/src" before mod_fax (at least I
> guess so). Currently the Makefile referred to libspandsp.la before it is
> compiled.
>
> regards
> helmut
>
>
>
> Michael Jerris schrieb:
>> make sure you have libtiff and libtiff dev packages installed then re-
>> configure freeswitch
>>
>> Mike
>>
>> On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:
>>
>> Hello,
>>
>> I tried to compile mod_fax today with trunk from yesterday. A 'make'
>> in
>> FS trunk directory led to an error saying that libspandsp.la wasn't
>> found in libs/spandsp/src. So I had to configure and compile (make)
>> spandsp manually before compiling FS.
>>
>> regards
>> helmut
>>
>>>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAkk+mCYACgkQ4tZeNddg3dxlPgCgpey84xCtTAD0GyiyDP3uPxpz
> SPgAnRJNO1s3n3xabGSbJYPtQmti2VKT
> =4Tja
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From regs at kinetix.gr  Tue Dec  9 08:38:21 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Tue, 09 Dec 2008 18:38:21 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>
	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
Message-ID: <493E9EFD.6040203@kinetix.gr>

That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).

In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.



Shelby Ramsey wrote:
> Hello,
>
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise 
> can be a little trying (a row in your CDR database could be over 100 
> fields long!).  
>
> The best option here is to catch the UUID's for the 2 call legs, 
> capture all SIP messaging, parse and dump the messaging, and then 
> correlate the calls from the CDR from there.  
>
> Much easier than trying to do it from FS ... and most folks want to 
> see SIP captures anyway (very broad set of tools to debug).  
>
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.
>
>
>
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
>
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be
>     introduced
>     to FS come from real life issues that we have faced during the
>     last few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
>
>
>
>
>     Michael Collins wrote:
>>     Thanks for your feedback. It definitely helps to know not only what
>>     you need FS to do but why you need it to do so.
>>
>>     Do you have FS in production right now? Just curious.
>>
>>     Thanks,
>>     MC
>>
>>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "I already added 2 patches for you right.  Just be clear about what you
>>>     want."
>>>
>>>     And I am grateful of that.
>>>
>>>     "it is protocol neutral, that's why it starts with sip_"
>>>
>>>     I didn't know that. I thought that the sip_ variables are protocol specific.
>>>     So one would expect there to be an iax_hangup_disposition,
>>>     woomera_hangup_disposition etc?
>>>
>>>     "Maybe you should beat around the bush less with your "requirements" for
>>>     your application you are expecting me to support for you."
>>>
>>>     I am just trying to gather statistics for my providers as I would with any
>>>     VoIP softswitch. (hangup causes per terminator per destination)
>>>     I don't think that this is a specific "application" rather than a general
>>>     necessity for VoIP carriers. It is also very useful for troubleshooting
>>>     purposes : when I look at my CDRs to find a call that I got a complain for,
>>>     I want to be able to tell if it was me or the provider who
>>>     hanged up and gave a specific hangup cause, so that I can troubleshoot the
>>>     issue better.
>>>
>>>     "Just be clear about what you want."
>>>
>>>     I want FS to reach that level of detailing and maturity in all aspects so
>>>     that it could be the softswitch of choice by any VoIP entrepreneur
>>>     (or hobbyist) and it is my strong belief that this can only be done by the
>>>     community giving feedback to the programmers about what
>>>     they find useful or not (i.e. experience from real-life situations). The
>>>     patches that you made the last few days were not intended for
>>>     me exclusively but for anyone that will face the same situations using FS.
>>>     If you want the community to stop sending feedback about
>>>     features/improvements you may as well close down this mailing list or just
>>>     use it as an announcement board.
>>>
>>>     I wish I was a c programmer and get involved with the project actively. But
>>>     I am not. And as far as I can tell most of the registered users
>>>     in this list aren't either. So they only way we can help is by testing and
>>>     suggesting.
>>>
>>>     Anthony Minessale wrote:
>>>
>>>     it is protocol neutral, that's why it starts with sip_
>>>
>>>     the variable can be any of:
>>>
>>>     send_bye
>>>     recv_bye
>>>     send_cancel
>>>     send_refuse
>>>
>>>
>>>     using that value you can determine the information you asked.  I answered
>>>     your specific question which was:
>>>     determining "which side hanged up".  Maybe you should beat around the bush
>>>     less with your "requirements" for your application you are expecting me to
>>>     support for you.
>>>
>>>     I already added 2 patches for you right.  Just be clear about what you want.
>>>
>>>
>>>
>>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     Not necessarily. For instance I got a "send_cancel" when the
>>>>     calling party hanged up before the other party could pick up.
>>>>     Also, shouldn't something like that be protocol/technology
>>>>     neutral?
>>>>
>>>>
>>>>
>>>>     Anthony Minessale wrote:
>>>>
>>>>     sip_hangup_disposition will be set to recv_bye on the side that was
>>>>     hungup.
>>>>
>>>>
>>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     Any updates about the "which side hanged up" potential variable?
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     Makes sense. I will look into this.
>>>>>     -MC
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>     I am sending this second email to request/suggest/enquire about something
>>>>>     relevant :
>>>>>
>>>>>     Wouldn't it be useful to know which end of a specific call leg send the
>>>>>     protocol
>>>>>     specific hangup cause? Otherwise it would be difficult to understand what
>>>>>     really happened.
>>>>>
>>>>>
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     I will do some research on this and let you know what I find out.
>>>>>     Question: are these internal calls or pstn or ?? Just curious about
>>>>>     your environment.
>>>>>
>>>>>     Thanks,
>>>>>     MC
>>>>>
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>
>>>>>
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>     ________________________________
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>         
>>     _______________________________________________
>>     Freeswitch-users mailing list
>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From regs at kinetix.gr  Tue Dec  9 08:39:57 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Tue, 09 Dec 2008 18:39:57 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>
	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
Message-ID: <493E9F5D.7020906@kinetix.gr>

That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).

In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.

Shelby Ramsey wrote:
> Hello,
>
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise 
> can be a little trying (a row in your CDR database could be over 100 
> fields long!).  
>
> The best option here is to catch the UUID's for the 2 call legs, 
> capture all SIP messaging, parse and dump the messaging, and then 
> correlate the calls from the CDR from there.  
>
> Much easier than trying to do it from FS ... and most folks want to 
> see SIP captures anyway (very broad set of tools to debug).  
>
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.
>
>
>
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
>
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be
>     introduced
>     to FS come from real life issues that we have faced during the
>     last few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
>
>
>
>
>     Michael Collins wrote:
>>     Thanks for your feedback. It definitely helps to know not only what
>>     you need FS to do but why you need it to do so.
>>
>>     Do you have FS in production right now? Just curious.
>>
>>     Thanks,
>>     MC
>>
>>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "I already added 2 patches for you right.  Just be clear about what you
>>>     want."
>>>
>>>     And I am grateful of that.
>>>
>>>     "it is protocol neutral, that's why it starts with sip_"
>>>
>>>     I didn't know that. I thought that the sip_ variables are protocol specific.
>>>     So one would expect there to be an iax_hangup_disposition,
>>>     woomera_hangup_disposition etc?
>>>
>>>     "Maybe you should beat around the bush less with your "requirements" for
>>>     your application you are expecting me to support for you."
>>>
>>>     I am just trying to gather statistics for my providers as I would with any
>>>     VoIP softswitch. (hangup causes per terminator per destination)
>>>     I don't think that this is a specific "application" rather than a general
>>>     necessity for VoIP carriers. It is also very useful for troubleshooting
>>>     purposes : when I look at my CDRs to find a call that I got a complain for,
>>>     I want to be able to tell if it was me or the provider who
>>>     hanged up and gave a specific hangup cause, so that I can troubleshoot the
>>>     issue better.
>>>
>>>     "Just be clear about what you want."
>>>
>>>     I want FS to reach that level of detailing and maturity in all aspects so
>>>     that it could be the softswitch of choice by any VoIP entrepreneur
>>>     (or hobbyist) and it is my strong belief that this can only be done by the
>>>     community giving feedback to the programmers about what
>>>     they find useful or not (i.e. experience from real-life situations). The
>>>     patches that you made the last few days were not intended for
>>>     me exclusively but for anyone that will face the same situations using FS.
>>>     If you want the community to stop sending feedback about
>>>     features/improvements you may as well close down this mailing list or just
>>>     use it as an announcement board.
>>>
>>>     I wish I was a c programmer and get involved with the project actively. But
>>>     I am not. And as far as I can tell most of the registered users
>>>     in this list aren't either. So they only way we can help is by testing and
>>>     suggesting.
>>>
>>>     Anthony Minessale wrote:
>>>
>>>     it is protocol neutral, that's why it starts with sip_
>>>
>>>     the variable can be any of:
>>>
>>>     send_bye
>>>     recv_bye
>>>     send_cancel
>>>     send_refuse
>>>
>>>
>>>     using that value you can determine the information you asked.  I answered
>>>     your specific question which was:
>>>     determining "which side hanged up".  Maybe you should beat around the bush
>>>     less with your "requirements" for your application you are expecting me to
>>>     support for you.
>>>
>>>     I already added 2 patches for you right.  Just be clear about what you want.
>>>
>>>
>>>
>>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     Not necessarily. For instance I got a "send_cancel" when the
>>>>     calling party hanged up before the other party could pick up.
>>>>     Also, shouldn't something like that be protocol/technology
>>>>     neutral?
>>>>
>>>>
>>>>
>>>>     Anthony Minessale wrote:
>>>>
>>>>     sip_hangup_disposition will be set to recv_bye on the side that was
>>>>     hungup.
>>>>
>>>>
>>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     Any updates about the "which side hanged up" potential variable?
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     Makes sense. I will look into this.
>>>>>     -MC
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>     I am sending this second email to request/suggest/enquire about something
>>>>>     relevant :
>>>>>
>>>>>     Wouldn't it be useful to know which end of a specific call leg send the
>>>>>     protocol
>>>>>     specific hangup cause? Otherwise it would be difficult to understand what
>>>>>     really happened.
>>>>>
>>>>>
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     I will do some research on this and let you know what I find out.
>>>>>     Question: are these internal calls or pstn or ?? Just curious about
>>>>>     your environment.
>>>>>
>>>>>     Thanks,
>>>>>     MC
>>>>>
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>
>>>>>
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>     ________________________________
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>         
>>     _______________________________________________
>>     Freeswitch-users mailing list
>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From vhatz at kinetix.gr  Tue Dec  9 08:42:54 2008
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Tue, 09 Dec 2008 18:42:54 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>
	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
Message-ID: <493EA00E.7070907@kinetix.gr>

Shelby Ramsey wrote:
> Hello,
> 
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise can 
> be a little trying (a row in your CDR database could be over 100 fields 
> long!).  
> 
> The best option here is to catch the UUID's for the 2 call legs, capture 
> all SIP messaging, parse and dump the messaging, and then correlate the 
> calls from the CDR from there.  
> 
> Much easier than trying to do it from FS ... and most folks want to see 
> SIP captures anyway (very broad set of tools to debug).  
> 
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.

That can certainly be an option, especially for debugging purposes.

However, under heavy load (imagine a few thousands of calls per hour, a 
few millions per day) logging and parsing all the SIP messages on file 
will be a problem.

Also, logging SIP messages is oriented to SIP only, when a more protocol 
agnostic approach could be followed. Plus, we would still need to parse 
a lot of text to extract the information that we need, while in a CDR 
(even a long one with many fields) we have a lot of information with a 
minimum hassle.

We've seen in production environments that excessive logging wastes I/O 
power and disk space, this is why (we at least) turn it on in our 
various systems only when we need it for troubleshooting, and 
immediately turn it off afterwards.

Additionally, a very long CDR is a lot less text to write on disk once, 
after the call is over, rather than writing many, whole packets during 
the duration of a call.

A 100-field-CDR on file could not be much of a problem, because usually 
these the raw CDR fields are rarely imported in a database in their 
entirety for billing or QoS analysis. A lot of the information which is 
not used directly & immediately for billing or QoS analysis remains on 
file in case needs to do basic troubleshooting in arrears. Granted, we 
would not have the same amount of information as with the written SIP 
messages, but it is useful nonetheless.

Of course, I need to stress that I write all this coming from the 
background of VoIP carriers. The above could apply well for typical & 
simple scenarios, where a call leg comes into FS and another calls leg 
comes out of it, which is what most carriers do.

If we need billing and QoS analysis for IVR's, queues, call transfers, 
etc, then yes, one-line CDRs would not do. In this case, logging whole 
packets could be a solution, although an event-based approach could be 
much better to cover all protocols/technologies (IAX, TDM cards, etc), IMHO.

Best regards,
Vlasis Hatzistavrou
Kinetix Tele.com Hellas Ltd.
Monastiriou 9 & Enotikon
54627
Thessaloniki
Greece
Tel.: +302310556134
Fax: +302310556134 (ext. 0)
GSM: +306977835653
e-mail: vhatz at kinetix.gr
http://www.kinetix.gr



> 
> 
> 
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs at kinetix.gr 
> <mailto:regs at kinetix.gr>> wrote:
> 
> 
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
> 
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be introduced
>     to FS come from real life issues that we have faced during the last
>     few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
> 
> 
> 
> 
>     Michael Collins wrote:
>>     Thanks for your feedback. It definitely helps to know not only what
>>     you need FS to do but why you need it to do so.
>>
>>     Do you have FS in production right now? Just curious.
>>
>>     Thanks,
>>     MC
>>
>>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "I already added 2 patches for you right.  Just be clear about what you
>>>     want."
>>>
>>>     And I am grateful of that.
>>>
>>>     "it is protocol neutral, that's why it starts with sip_"
>>>
>>>     I didn't know that. I thought that the sip_ variables are protocol specific.
>>>     So one would expect there to be an iax_hangup_disposition,
>>>     woomera_hangup_disposition etc?
>>>
>>>     "Maybe you should beat around the bush less with your "requirements" for
>>>     your application you are expecting me to support for you."
>>>
>>>     I am just trying to gather statistics for my providers as I would with any
>>>     VoIP softswitch. (hangup causes per terminator per destination)
>>>     I don't think that this is a specific "application" rather than a general
>>>     necessity for VoIP carriers. It is also very useful for troubleshooting
>>>     purposes : when I look at my CDRs to find a call that I got a complain for,
>>>     I want to be able to tell if it was me or the provider who
>>>     hanged up and gave a specific hangup cause, so that I can troubleshoot the
>>>     issue better.
>>>
>>>     "Just be clear about what you want."
>>>
>>>     I want FS to reach that level of detailing and maturity in all aspects so
>>>     that it could be the softswitch of choice by any VoIP entrepreneur
>>>     (or hobbyist) and it is my strong belief that this can only be done by the
>>>     community giving feedback to the programmers about what
>>>     they find useful or not (i.e. experience from real-life situations). The
>>>     patches that you made the last few days were not intended for
>>>     me exclusively but for anyone that will face the same situations using FS.
>>>     If you want the community to stop sending feedback about
>>>     features/improvements you may as well close down this mailing list or just
>>>     use it as an announcement board.
>>>
>>>     I wish I was a c programmer and get involved with the project actively. But
>>>     I am not. And as far as I can tell most of the registered users
>>>     in this list aren't either. So they only way we can help is by testing and
>>>     suggesting.
>>>
>>>     Anthony Minessale wrote:
>>>
>>>     it is protocol neutral, that's why it starts with sip_
>>>
>>>     the variable can be any of:
>>>
>>>     send_bye
>>>     recv_bye
>>>     send_cancel
>>>     send_refuse
>>>
>>>
>>>     using that value you can determine the information you asked.  I answered
>>>     your specific question which was:
>>>     determining "which side hanged up".  Maybe you should beat around the bush
>>>     less with your "requirements" for your application you are expecting me to
>>>     support for you.
>>>
>>>     I already added 2 patches for you right.  Just be clear about what you want.
>>>
>>>
>>>
>>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     Not necessarily. For instance I got a "send_cancel" when the
>>>>     calling party hanged up before the other party could pick up.
>>>>     Also, shouldn't something like that be protocol/technology
>>>>     neutral?
>>>>
>>>>
>>>>
>>>>     Anthony Minessale wrote:
>>>>
>>>>     sip_hangup_disposition will be set to recv_bye on the side that was
>>>>     hungup.
>>>>
>>>>
>>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     Any updates about the "which side hanged up" potential variable?
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     Makes sense. I will look into this.
>>>>>     -MC
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>     I am sending this second email to request/suggest/enquire about something
>>>>>     relevant :
>>>>>
>>>>>     Wouldn't it be useful to know which end of a specific call leg send the
>>>>>     protocol
>>>>>     specific hangup cause? Otherwise it would be difficult to understand what
>>>>>     really happened.
>>>>>
>>>>>
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     I will do some research on this and let you know what I find out.
>>>>>     Question: are these internal calls or pstn or ?? Just curious about
>>>>>     your environment.
>>>>>
>>>>>     Thanks,
>>>>>     MC
>>>>>
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>
>>>>>
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>     ________________________________
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>         
>>     _______________________________________________
>>     Freeswitch-users mailing list
>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
> 
> 
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
> 
> 
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From woof at nortel.com  Tue Dec  9 08:45:03 2008
From: woof at nortel.com (Andy Spitzer)
Date: Tue, 09 Dec 2008 11:45:03 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
Message-ID: <op.ulwl5dbessieo1@woof.us.nortel.com>

Woof!

It appears that FreeSWITCH writes 

   freeswitch.history
   freeswitch.log
   freeswitch.pid
   freeswitch.xml.fsxml

to the -log directory.  

Is there a way to put the files other than freeswitch.log into the -db directory instead?

In my environment we archive and rotate everything in the log directory (which includes logs beside FreeSWITCH's), and these other FreeSWITCH files are getting rotated.  Yeah, I can explicitly exclude them, but to me it seems those really belong in the -db directory anyway, as they are inherently data needed for the current executable of FreeSWITCH, and not logs.

--Woof!


From mrjoebain at gmail.com  Tue Dec  9 08:53:30 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 9 Dec 2008 16:53:30 +0000
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <8F96510A-1A1E-45E9-A206-FDA66CAEA06F@jerris.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
	<1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
	<748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>
	<748d53500812090610t70a11a07u594541a8e132a9d3@mail.gmail.com>
	<8F96510A-1A1E-45E9-A206-FDA66CAEA06F@jerris.com>
Message-ID: <748d53500812090853v2339ac3x2ca77446884eab9@mail.gmail.com>

2008/12/9 Michael Jerris <mike at jerris.com>

>
>  On Dec 9, 2008, at 9:10 AM, Joe Bain wrote:
>
>  Ok I have been testing more and I have reduced my problem to a pretty
> short and simple Lua script. I've posted it at
> http://pastebin.freeswitch.org/6373 and this gets called straight from the
> dialplan. From my experience so far it only exits after a caller hangup
> about 1 in 10 times. Most of the time it continues to loop until I do
> 'hupall'.
>
> Thanks in advance if anyone can solve this or offer any advice.
>
> Joe
>
> 2008/12/9 Joe Bain <mrjoebain at gmail.com>
>
>> 2008/12/9 Ivan C Myrvold <ivan at myrvold.org>
>>
>>  Did you read carefully when asked to provide login and password? The
>>> login and password is there, don't use your own freeswitch login.
>>>
>>> Ivan
>>>
>>>  Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>>>
>>>  On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
>>> > Hi,
>>> >
>>> > I'm writing an IVR in Lua and am having problems dealing with hangups
>>> > cleanly. Very often session:ready() reports true long after I have hung
>>> up
>>> > and the hangup hook function I have set doesn't get called either. It
>>> seems
>>> > to report that the session is active indefinitely in some cases where a
>>> loop
>>> > keeps trying to get some dtmf key presses. Is there any trick to using
>>> > session:ready() or the hangup hook that I might have missed?
>>> >
>>> > On a slightly related point I can't seem to access the session
>>> properties,
>>> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
>>> >
>>>
>>>
>        Joe,
>>>
>>>>
>>>> A few questions... what svn rev are you running? Which operating
>>>> system? Finally, is it possible for you to put your dialplan and Lua
>>>> script up at pastebin.freeswitch.org?
>>>>
>>>> Thanks,
>>>> MC
>>>>
>>> Hi,
>>>
>>> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I can
>>> post the dialplan and lua script though at the moment I can't seem to log in
>>> to the pastebin, I just became a member on the freeswitch homepage but the
>>> pass/username isn't being accepted.
>>>
>>> Joe
>>>
>>>
>> Ah, I should have read more carefully!
>>
>> The dialplan is here <http://pastebin.freeswitch.org/6372> and the two
>> important lua scripts are here <http://pastebin.freeswitch.org/6371> and
>> here <http://pastebin.freeswitch.org/6370>, the first calls the second. I
>> didn't include all the Lua script as the problem appears right at the start
>> (as well as throughout) if the user hangs up when the IVR is speaking
>> (asking for an id number) then it seems to never get a hangup and loops
>> trying to get the id number.
>>
>> Joe
>>
>
> We just tested this with current svn trunk and it appears to work fine,
> could you try updating and see if it is still a problem for you
>
> Mike
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
I have to install FS on our server soon so when I do I'll see if the problem
is still there. Though I'm testing on Vista and the server won't be running
that so if the problem doesn't reappear it may not be conclusive. If I have
any spare time I'll try a reinstall on my test machine but I probably won't
unfortunately, my contract ends on wednesday at the company I'm working for.

Thanks for your help.

Joe
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From anthony.minessale at gmail.com  Tue Dec  9 09:08:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Dec 2008 11:08:51 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493EA00E.7070907@kinetix.gr>
References: <49391D40.6050103@kinetix.gr> <493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
	<493D2B9B.6050205@kinetix.gr>
	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
	<493E2A74.7010502@kinetix.gr>
	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>
	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
	<493EA00E.7070907@kinetix.gr>
Message-ID: <191c3a030812090908kca14327v821d78998e8b50b6@mail.gmail.com>

see this is better.

That's why I asked you to be more specific about what you want because the
tiny back and
forth questions were not exposing your intent or needs at all.  I answer
every email I can and when threads start to grow without getting to the
point i start to get frustrated.

Now that you have opened up the discussion you have more people chiming in
on the topic.

Yes the sip_* variables are unique to SIP and the one that says
proto_specific are all
done per implementation.

If you would like to suggest a list of standard variables you think apply to
all calls
or something you feel is missing, we can consider injecting them into the
code.




On Tue, Dec 9, 2008 at 10:42 AM, Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr
> wrote:

> Shelby Ramsey wrote:
> > Hello,
> >
> > This is just my 2 cents ... but my experience has been that trying to
> > catch all of the various variables (i.e. from XML_CDR) or otherwise can
> > be a little trying (a row in your CDR database could be over 100 fields
> > long!).
> >
> > The best option here is to catch the UUID's for the 2 call legs, capture
> > all SIP messaging, parse and dump the messaging, and then correlate the
> > calls from the CDR from there.
> >
> > Much easier than trying to do it from FS ... and most folks want to see
> > SIP captures anyway (very broad set of tools to debug).
> >
> > Measuring things like ASR, PDD, etc in my opinion is much easier from
> > the raw messaging than trying to do something with FS CDR records.
>
> That can certainly be an option, especially for debugging purposes.
>
> However, under heavy load (imagine a few thousands of calls per hour, a
> few millions per day) logging and parsing all the SIP messages on file
> will be a problem.
>
> Also, logging SIP messages is oriented to SIP only, when a more protocol
> agnostic approach could be followed. Plus, we would still need to parse
> a lot of text to extract the information that we need, while in a CDR
> (even a long one with many fields) we have a lot of information with a
> minimum hassle.
>
> We've seen in production environments that excessive logging wastes I/O
> power and disk space, this is why (we at least) turn it on in our
> various systems only when we need it for troubleshooting, and
> immediately turn it off afterwards.
>
> Additionally, a very long CDR is a lot less text to write on disk once,
> after the call is over, rather than writing many, whole packets during
> the duration of a call.
>
> A 100-field-CDR on file could not be much of a problem, because usually
> these the raw CDR fields are rarely imported in a database in their
> entirety for billing or QoS analysis. A lot of the information which is
> not used directly & immediately for billing or QoS analysis remains on
> file in case needs to do basic troubleshooting in arrears. Granted, we
> would not have the same amount of information as with the written SIP
> messages, but it is useful nonetheless.
>
> Of course, I need to stress that I write all this coming from the
> background of VoIP carriers. The above could apply well for typical &
> simple scenarios, where a call leg comes into FS and another calls leg
> comes out of it, which is what most carriers do.
>
> If we need billing and QoS analysis for IVR's, queues, call transfers,
> etc, then yes, one-line CDRs would not do. In this case, logging whole
> packets could be a solution, although an event-based approach could be
> much better to cover all protocols/technologies (IAX, TDM cards, etc),
> IMHO.
>
> Best regards,
> Vlasis Hatzistavrou
> Kinetix Tele.com Hellas Ltd.
> Monastiriou 9 & Enotikon
> 54627
> Thessaloniki
> Greece
> Tel.: +302310556134
> Fax: +302310556134 (ext. 0)
> GSM: +306977835653
> e-mail: vhatz at kinetix.gr
> http://www.kinetix.gr
>
>
>
> >
> >
> >
> > On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs at kinetix.gr
> > <mailto:regs at kinetix.gr>> wrote:
> >
> >
> >         We are currently in the migration process from our
> >     current system to a FS based setup. We are in the process of
> >     adapting our billing and routing to FS. All the  CDRs (and variables)
> >     related issues that we have been discussing on this mailing list
> >     come from the need to extract the same level of information from FS
> as
> >     we do with our current closed source proprietary system. So, we
> >     chose FS because of the versatility it provides in every aspect
> (event
> >     handling, config implementation etc.) and we strongly believe that
> all
> >     these additions/fixes would be beneficial to many potential FS users.
> >
> >         We are at your disposal for more details in case you need
> >     more information about what exactly we are trying to do. Basically,
> >     our approach is from the VoIP carrier's point of view rather than the
> >     PBX user's/implementor's. So, the details that we asked to be
> introduced
> >     to FS come from real life issues that we have faced during the last
> >     few years
> >     with various platforms and troubleshooting experiences with other
> >     VoIP carriers.
> >
> >
> >
> >
> >     Michael Collins wrote:
> >>     Thanks for your feedback. It definitely helps to know not only what
> >>     you need FS to do but why you need it to do so.
> >>
> >>     Do you have FS in production right now? Just curious.
> >>
> >>     Thanks,
> >>     MC
> >>
> >>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
> >>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
> >>
> >>>     "I already added 2 patches for you right.  Just be clear about what
> you
> >>>     want."
> >>>
> >>>     And I am grateful of that.
> >>>
> >>>     "it is protocol neutral, that's why it starts with sip_"
> >>>
> >>>     I didn't know that. I thought that the sip_ variables are protocol
> specific.
> >>>     So one would expect there to be an iax_hangup_disposition,
> >>>     woomera_hangup_disposition etc?
> >>>
> >>>     "Maybe you should beat around the bush less with your
> "requirements" for
> >>>     your application you are expecting me to support for you."
> >>>
> >>>     I am just trying to gather statistics for my providers as I would
> with any
> >>>     VoIP softswitch. (hangup causes per terminator per destination)
> >>>     I don't think that this is a specific "application" rather than a
> general
> >>>     necessity for VoIP carriers. It is also very useful for
> troubleshooting
> >>>     purposes : when I look at my CDRs to find a call that I got a
> complain for,
> >>>     I want to be able to tell if it was me or the provider who
> >>>     hanged up and gave a specific hangup cause, so that I can
> troubleshoot the
> >>>     issue better.
> >>>
> >>>     "Just be clear about what you want."
> >>>
> >>>     I want FS to reach that level of detailing and maturity in all
> aspects so
> >>>     that it could be the softswitch of choice by any VoIP entrepreneur
> >>>     (or hobbyist) and it is my strong belief that this can only be done
> by the
> >>>     community giving feedback to the programmers about what
> >>>     they find useful or not (i.e. experience from real-life
> situations). The
> >>>     patches that you made the last few days were not intended for
> >>>     me exclusively but for anyone that will face the same situations
> using FS.
> >>>     If you want the community to stop sending feedback about
> >>>     features/improvements you may as well close down this mailing list
> or just
> >>>     use it as an announcement board.
> >>>
> >>>     I wish I was a c programmer and get involved with the project
> actively. But
> >>>     I am not. And as far as I can tell most of the registered users
> >>>     in this list aren't either. So they only way we can help is by
> testing and
> >>>     suggesting.
> >>>
> >>>     Anthony Minessale wrote:
> >>>
> >>>     it is protocol neutral, that's why it starts with sip_
> >>>
> >>>     the variable can be any of:
> >>>
> >>>     send_bye
> >>>     recv_bye
> >>>     send_cancel
> >>>     send_refuse
> >>>
> >>>
> >>>     using that value you can determine the information you asked.  I
> answered
> >>>     your specific question which was:
> >>>     determining "which side hanged up".  Maybe you should beat around
> the bush
> >>>     less with your "requirements" for your application you are
> expecting me to
> >>>     support for you.
> >>>
> >>>     I already added 2 patches for you right.  Just be clear about what
> you want.
> >>>
> >>>
> >>>
> >>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <
> regs at kinetix.gr> <mailto:regs at kinetix.gr>
> >>>     wrote:
> >>>
> >>>>     Not necessarily. For instance I got a "send_cancel" when the
> >>>>     calling party hanged up before the other party could pick up.
> >>>>     Also, shouldn't something like that be protocol/technology
> >>>>     neutral?
> >>>>
> >>>>
> >>>>
> >>>>     Anthony Minessale wrote:
> >>>>
> >>>>     sip_hangup_disposition will be set to recv_bye on the side that
> was
> >>>>     hungup.
> >>>>
> >>>>
> >>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <
> regs at kinetix.gr> <mailto:regs at kinetix.gr>
> >>>>     wrote:
> >>>>
> >>>>>     Any updates about the "which side hanged up" potential variable?
> >>>>>
> >>>>>     Michael S Collins wrote:
> >>>>>
> >>>>>     Makes sense. I will look into this.
> >>>>>     -MC
> >>>>>
> >>>>>
> >>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <
> regs at kinetix.gr> <mailto:regs at kinetix.gr>
> >>>>>     wrote:
> >>>>>
> >>>>>     I am sending this second email to request/suggest/enquire about
> something
> >>>>>     relevant :
> >>>>>
> >>>>>     Wouldn't it be useful to know which end of a specific call leg
> send the
> >>>>>     protocol
> >>>>>     specific hangup cause? Otherwise it would be difficult to
> understand what
> >>>>>     really happened.
> >>>>>
> >>>>>
> >>>>>
> >>>>>     Michael S Collins wrote:
> >>>>>
> >>>>>     I will do some research on this and let you know what I find out.
> >>>>>     Question: are these internal calls or pstn or ?? Just curious
> about
> >>>>>     your environment.
> >>>>>
> >>>>>     Thanks,
> >>>>>     MC
> >>>>>
> >>>>>
> >>>>>
> >>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <
> regs at kinetix.gr> <mailto:regs at kinetix.gr>
> >>>>>     wrote:
> >>>>>
> >>>>>
> >>>>>
> >>>>>     The proto_specific_hangup_cause is missing on one of the two
> >>>>>     call legs. When the caller hangs up it is missing from the a-leg
> CDR.
> >>>>>     When the callee hangs up it is missing from the b-leg CDR. Is
> this
> >>>>>     nornal?
> >>>>>
> >>>>>     And another question : what piece of info could inform me about
> who
> >>>>>     hanged up?
> >>>>>
> >>>>>
> >>>>>     --
> >>>>>     -------------------------------------------
> >>>>>     Apostolos Pantsiopoulos
> >>>>>     Kinetix Tele.com R & D
> >>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>>>     -------------------------------------------
> >>>>>
> >>>>>
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>     --
> >>>>>     -------------------------------------------
> >>>>>     Apostolos Pantsiopoulos
> >>>>>     Kinetix Tele.com R & D
> >>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>>>     -------------------------------------------
> >>>>>
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>     ________________________________
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>     --
> >>>>>     -------------------------------------------
> >>>>>     Apostolos Pantsiopoulos
> >>>>>     Kinetix Tele.com R & D
> >>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>>>     -------------------------------------------
> >>>>>
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>     --
> >>>>     Anthony Minessale II
> >>>>
> >>>>     FreeSWITCH http://www.freeswitch.org/
> >>>>     ClueCon http://www.cluecon.com/
> >>>>
> >>>>     AIM: anthm
> >>>>     MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com><mailto:
> MSN:anthony_minessale at hotmail.com>
> >>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com><mailto:
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> >>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>>>
> >>>>     FreeSWITCH Developer Conference
> >>>>     sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:
> guest at conference.freeswitch.org/888>
> >>>>     googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org><mailto:
> googletalk:conf+888 at conference.freeswitch.org<conf%2B888 at conference.freeswitch.org>
> >
> >>>>     pstn:213-799-1400
> >>>>
> >>>>     ________________________________
> >>>>     _______________________________________________
> >>>>     Freeswitch-users mailing list
> >>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>     http://www.freeswitch.org
> >>>>
> >>>>
> >>>>     --
> >>>>     -------------------------------------------
> >>>>     Apostolos Pantsiopoulos
> >>>>     Kinetix Tele.com R & D
> >>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>>     -------------------------------------------
> >>>>
> >>>>     _______________________________________________
> >>>>     Freeswitch-users mailing list
> >>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>     http://www.freeswitch.org
> >>>>
> >>>>
> >>>     --
> >>>     Anthony Minessale II
> >>>
> >>>     FreeSWITCH http://www.freeswitch.org/
> >>>     ClueCon http://www.cluecon.com/
> >>>
> >>>     AIM: anthm
> >>>     MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com><mailto:
> MSN:anthony_minessale at hotmail.com>
> >>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com><mailto:
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> >>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>>
> >>>     FreeSWITCH Developer Conference
> >>>     sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:
> guest at conference.freeswitch.org/888>
> >>>     googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org><mailto:
> googletalk:conf+888 at conference.freeswitch.org<conf%2B888 at conference.freeswitch.org>
> >
> >>>     pstn:213-799-1400
> >>>
> >>>     ________________________________
> >>>     _______________________________________________
> >>>     Freeswitch-users mailing list
> >>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>     http://www.freeswitch.org
> >>>
> >>>
> >>>     --
> >>>     -------------------------------------------
> >>>     Apostolos Pantsiopoulos
> >>>     Kinetix Tele.com R & D
> >>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>     -------------------------------------------
> >>>
> >>>     _______________________________________________
> >>>     Freeswitch-users mailing list
> >>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>     http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>     _______________________________________________
> >>     Freeswitch-users mailing list
> >>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     http://www.freeswitch.org
> >>
> >
> >
> >     --
> >     -------------------------------------------
> >     Apostolos Pantsiopoulos
> >     Kinetix Tele.com R & D
> >     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >     -------------------------------------------
> >
> >
> >     _______________________________________________
> >     Freeswitch-users mailing list
> >     Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From erick at junctionnetworks.com  Tue Dec  9 10:06:38 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 13:06:38 -0500
Subject: [Freeswitch-users]  Sending SIP calls via outbound proxy
References: 493DC658.8020305@junctionnetworks.com
Message-ID: <493EB3AE.1090109@junctionnetworks.com>

Hi Brian,

Thanks for the reply, but I still don't think that answers my original
question.  I'm trying to get FS to act simply as a UAC in this
instance, what I want is for FS to proxy ALL outbound calls through my
proxy server at foo.com.

So when FS originates a call to alice at bar.com I want the signaling path
to be set up as:

FreeSwitch ---> proxy.foo.com ---> alice at bar.com

I found this thread: 
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/008582.html
but I still can't seem to get FS to stop resolving domain bar.com on 
it's own, even when I set the
sip_invite_domain variable like so:
 originate 
{sip_invite_domain='proxy.foo.com'}sofia/external/alice at bar.com &echo()

That is how I ended up using the "originate 
sofia/gateway/proxy/alice at bar.com &echo()" command.
While I understand that's wrong, I don't know what the right config/cmd 
is to accomplish my task.

Any other help is much appreciated.

Thanks,

Erick


>
> First example is WRONG you don't dial via a gateway that way.  If you  
> wish to dial alice at bar.com then try sofia/internal/alice at bar.com 
> as  
> you don't require a gateway to call alice right?
>
> /b
>
> On Dec 8, 2008, at 7:14 PM, Erick Johnson wrote:
>
> > Here is the command that I'm trying to use that behaves unexpectedly:
> > originate sofia/gateway/proxy/alice at bar.com &echo()
> >
> > However this command produces the results I'm expecting:
> > originate sofia/gateway/proxy/15551234 &echo()
>

-- 
Erick Johnson

sip/email erick at junctionnetworks.com
1-800-801-3381 x7006
Software Engineer
Junction Networks



From brian at freeswitch.org  Tue Dec  9 10:15:47 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 12:15:47 -0600
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493EB3AE.1090109@junctionnetworks.com>
References: 493DC658.8020305@junctionnetworks.com
	<493EB3AE.1090109@junctionnetworks.com>
Message-ID: <E5B2EB7B-A125-4861-A23D-2051242E0C1D@freeswitch.org>

Can you clarify why you need a gateway?  Is the far side going to  
challenge us and request authentication credentials?

So you want us to not resolve the domain of the target at all in any  
way?  That kinda breaks the rules because you should always check the  
NAPTR's and SRV and resolve to the target in that manner its a  
requirement.  If you want to force things to a proxy and let the proxy  
on the far side do the work then you do this:

sofia/profile/alice at bar.com;fs_path=proxy.foo.com

/b



On Dec 9, 2008, at 12:06 PM, Erick Johnson wrote:

> Hi Brian,
>
> Thanks for the reply, but I still don't think that answers my original
> question.  I'm trying to get FS to act simply as a UAC in this
> instance, what I want is for FS to proxy ALL outbound calls through my
> proxy server at foo.com.
>
> So when FS originates a call to alice at bar.com I want the signaling  
> path
> to be set up as:
>
> FreeSwitch ---> proxy.foo.com ---> alice at bar.com
>
> I found this thread:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/008582.html
> but I still can't seem to get FS to stop resolving domain bar.com on
> it's own, even when I set the
> sip_invite_domain variable like so:
> originate
> {sip_invite_domain='proxy.foo.com'}sofia/external/alice at bar.com  
> &echo()
>
> That is how I ended up using the "originate
> sofia/gateway/proxy/alice at bar.com &echo()" command.
> While I understand that's wrong, I don't know what the right config/ 
> cmd
> is to accomplish my task.
>
> Any other help is much appreciated.
>
> Thanks,
>
> Erick

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From erick at junctionnetworks.com  Tue Dec  9 11:11:26 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 14:11:26 -0500
Subject: [Freeswitch-users]  Sending SIP calls via outbound proxy
References: 493EB3AE.1090109@junctionnetworks.com
Message-ID: <493EC2DE.2090609@junctionnetworks.com>

Thanks Brian,

I never did want to use a gateway - I was just lost on how to force FS 
to use a proxy.
fs_path seems to be what I'm looking for.  However now what I run my
originate command the channel gets terminated before FS even sends out
a packet.  The api call completes with NORMAL_UNSPECIFIED
termination cause.

API CALL 
[originate(sofia/external/erick at ejjohnson.org;fs_path=proxy.foo.net 
&echo())] output:
-ERR NORMAL_UNSPECIFIED

Looking at the logs the reason as to why it's been termintated isn't cleear
to me.  Any thoughts?

Here is the pastebin for the log http://pastebin.freeswitch.org/6378

Thanks again

E






> Can you clarify why you need a gateway?  Is the far side going to  
> challenge us and request authentication credentials?
>
> So you want us to not resolve the domain of the target at all in any  
> way?  That kinda breaks the rules because you should always check the  
> NAPTR's and SRV and resolve to the target in that manner its a  
> requirement.  If you want to force things to a proxy and let the proxy  
> on the far side do the work then you do this:
>
> sofia/profile/alice at bar.com;fs_path=proxy.foo.com
>
> /b
>
>


From brian at freeswitch.org  Tue Dec  9 11:17:41 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 13:17:41 -0600
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493EC2DE.2090609@junctionnetworks.com>
References: 493EB3AE.1090109@junctionnetworks.com
	<493EC2DE.2090609@junctionnetworks.com>
Message-ID: <71B232D7-BC0C-4028-B768-56960285C85E@freeswitch.org>

I think you need to '' the sofia uri

/b

On Dec 9, 2008, at 1:11 PM, Erick Johnson wrote:

> Looking at the logs the reason as to why it's been termintated isn't  
> cleear
> to me.  Any thoughts?



From erick at junctionnetworks.com  Tue Dec  9 11:41:55 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 14:41:55 -0500
Subject: [Freeswitch-users]  Sending SIP calls via outbound proxy
References: 493EC2DE.2090609@junctionnetworks.com
Message-ID: <493ECA03.2030106@junctionnetworks.com>

Both:

originate sofia/external/'erick at ejjohnson.org;fs_path=proxybeta.foo.net' 
&echo()
originate sofia/external/erick at ejjohnson.org;fs_path=proxybeta.foo.net 
&echo()

produce the exact same result & log

:(

> * I think you need to '' the sofia uri /b 



From msc at freeswitch.org  Tue Dec  9 11:52:59 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 11:52:59 -0800
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493ECA03.2030106@junctionnetworks.com>
References: <493ECA03.2030106@junctionnetworks.com>
Message-ID: <87f2f3b90812091152j43d9659agadf3eba00554c8b2@mail.gmail.com>

What SVN rev are you running? Also, could you do a SIP trace?
TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
Pastebin the output of that and we'll take it from there.
-MC

On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
<erick at junctionnetworks.com> wrote:
> Both:
>
> originate sofia/external/'erick at ejjohnson.org;fs_path=proxybeta.foo.net'
> &echo()
> originate sofia/external/erick at ejjohnson.org;fs_path=proxybeta.foo.net
> &echo()
>
> produce the exact same result & log
>
> :(
>
>> * I think you need to '' the sofia uri /b
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Dec  9 11:56:27 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 13:56:27 -0600
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <87f2f3b90812091152j43d9659agadf3eba00554c8b2@mail.gmail.com>
References: <493ECA03.2030106@junctionnetworks.com>
	<87f2f3b90812091152j43d9659agadf3eba00554c8b2@mail.gmail.com>
Message-ID: <06E304A9-C9D0-45AD-A24C-1D438CEB93C2@freeswitch.org>

originate 'sofia/internal/brian at bkw.org;fs_path=bob.com' &echo()

/b

On Dec 9, 2008, at 1:52 PM, Michael Collins wrote:

> What SVN rev are you running? Also, could you do a SIP trace?
> TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
> Pastebin the output of that and we'll take it from there.
> -MC
>
> On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
> <erick at junctionnetworks.com> wrote:
>> Both:
>>
>> originate sofia/ 
>> external/'erick at ejjohnson.org;fs_path=proxybeta.foo.net'
>> &echo()
>> originate sofia/external/ 
>> erick at ejjohnson.org;fs_path=proxybeta.foo.net
>> &echo()
>>
>> produce the exact same result & log
>>
>> :(
>>
>>> * I think you need to '' the sofia uri /b
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From erick at junctionnetworks.com  Tue Dec  9 12:47:08 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 15:47:08 -0500
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
References: 493ECA03.2030106@junctionnetworks.com
Message-ID: <493ED94C.9080508@junctionnetworks.com>

I'm running latest trunk - Revision: 10682

I've been doing an ngrep on my external freeswitch SIP port and FS
is not sending any SIP packets anywhere when I run the following command.
Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are logged.

originate 
'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' &echo()

Also, just to be clear, when I remove ";fs_path=..." from the command 
above a call
is set up normally to erick at ejjohnson.org and the SIP packets are logged 
to console.

Thanks guys.

> What SVN rev are you running? Also, could you do a SIP trace?
> TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
> Pastebin the output of that and we'll take it from there.
> -MC
>
> On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
> <erick at junctionnetworks.com> wrote:
> > Both:
> >
> > originate sofia/external/'erick at 
> ejjohnson.org;fs_path=proxybeta.foo.net'
> > &echo()
> > originate sofia/external/erick at 
> ejjohnson.org;fs_path=proxybeta.foo.net
> > &echo()
> >
> > produce the exact same result & log
> >
> > :(
> >
> >> * I think you need to '' the sofia uri /b
> >
> >



From erick at junctionnetworks.com  Tue Dec  9 12:49:42 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 15:49:42 -0500
Subject: [Freeswitch-users]  Sending SIP calls via outbound proxy
Message-ID: <493ED9E6.3000803@junctionnetworks.com>


i forgot to give you the pastebin URL
http://pastebin.freeswitch.org/6379

>
> I'm running latest trunk - Revision: 10682
>
> I've been doing an ngrep on my external freeswitch SIP port and FS
> is not sending any SIP packets anywhere when I run the following command.
> Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are 
> logged.
>
> originate
> 'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' 
> &echo()
>
> Also, just to be clear, when I remove ";fs_path=..." from the command
> above a call
> is set up normally to erick at ejjohnson.org and the SIP packets are 
> logged
> to console.
>
> Thanks guys.
>
> > What SVN rev are you running? Also, could you do a SIP trace?
> > TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
> > Pastebin the output of that and we'll take it from there.
> > -MC
> >
> > On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
> > <erick at junctionnetworks.com> wrote:
> > > Both:
> > >
> > > originate sofia/external/'erick at
> > ejjohnson.org;fs_path=proxybeta.foo.net'
> > > &echo()
> > > originate sofia/external/erick at
> > ejjohnson.org;fs_path=proxybeta.foo.net
> > > &echo()
> > >
> > > produce the exact same result & log
> > >
> > > :(
> > >
> > >> * I think you need to '' the sofia uri /b
> > >
> > >
>
>

-- 
Erick Johnson

sip/email erick at junctionnetworks.com
1-800-801-3381 x7006
Software Engineer
Junction Networks



From gilbertandrew at me.com  Tue Dec  9 12:54:01 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 09 Dec 2008 15:54:01 -0500
Subject: [Freeswitch-users] Question Regarding ASR/TTS and CMU OSS Projects
Message-ID: <BFC0BBD3-D9A7-48DA-A8CD-E8448B75A9F2@me.com>

Curious if anyone has practical real world input on training CMU based  
ASR engines (Sphinx, PocketSphinx) and / or creating and tuning voices  
for the TTS related components.

Just trying to understand how hard it is, what the realistic gap is to  
use these tools in real world applications.




From frank at impactfax.com  Tue Dec  9 14:35:47 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 9 Dec 2008 17:35:47 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <4CF7C22E-64D4-4EEF-8153-4999E06E3A9F@me.com>
Message-ID: <09aa01c95a4e$7b7a6ea0$33014c0a@ws4>

On our last bill, the carrier said we had 27% short duration calls
(maybe they are wrong but it was on the bill).  It is definitely not
call center. But these callers hangup as soon as they hear answer
machine or most of the time a ring back tone from cell phone.  This
class of caller will call a cell phone, hear the ring back, hangup right
away and then call back another 2 minutes later and repeat the cycle.
 
So, if I have to make it work the way I suggested (hold the connection
open for at least the minimum time, how might you suggest I do it in the
dial plan?
 
-----Original Message-----
 
Don't want the tone to be wrong here, but this makes no sense. Carriers
surcharge like this precisely to guard against call center, predictive
and other mass outbound calling scenarios.
 
It just doesn't make since, math wise, that individuals hanging up on
voice mail are going to significantly impact overall ACD stats, etc. So
unless you have a very strange set of use cases or are pushing another
category of traffic (ie call center) that skews you overall relationship
with the carrier - I would go back and re-negotiate your arrangement.
 
Yes, FS is a b2bua and all is possible. But it is probably a better use
of time to approach this as a business issue.
 
My 2 cents.
 
 
On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:



How can FS force a Minimum call duration for a FS caller (someone
calling out of FS)? 
 
We have a carrier that penalizes us with a surcharge for short duration
calls (sound familiar?).
 
So when a FS caller (not a call center or predictive dialer) calls a
cell phone and gets a ring tone or calls an answering machine, the FS
caller hangs up because they do not want to leave a message.  But they
do this in less then a few seconds after the call is answered.  This
becomes a short duration call and bang the surcharge applies.  It is
actually cheaper to pay for a longer call time (6 seconds in this case)
and avoid the short duration surcharge.  But the FS caller does not know
this.
 
So, how can FS hold the connection to the called party open for at least
the minimum amount of time I need to avoid the short call charge. even
though my FS caller has already hung up the phone on his end?  I would
like to do this in the xml dialplanif possible.
 
Thanks
 
-Frank
 
 
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From msc at freeswitch.org  Tue Dec  9 15:53:39 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 15:53:39 -0800
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <09aa01c95a4e$7b7a6ea0$33014c0a@ws4>
References: <4CF7C22E-64D4-4EEF-8153-4999E06E3A9F@me.com>
	<09aa01c95a4e$7b7a6ea0$33014c0a@ws4>
Message-ID: <87f2f3b90812091553s561b6b48kda080b1bee0dd775@mail.gmail.com>

Can you paste your dialplan entry here? I have some thoughts but it
would be better if I knew what you were doing before I go any further.
-MC



On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <frank at impactfax.com> wrote:
> On our last bill, the carrier said we had 27% short duration calls (maybe
> they are wrong but it was on the bill).  It is definitely not call center.
> But these callers hangup as soon as they hear answer machine or most of the
> time a ring back tone from cell phone.  This class of caller will call a
> cell phone, hear the ring back, hangup right away and then call back another
> 2 minutes later and repeat the cycle.
>
>
>
> So, if I have to make it work the way I suggested (hold the connection open
> for at least the minimum time, how might you suggest I do it in the dial
> plan?
>
>
>
> -----Original Message-----
>
>
> Don't want the tone to be wrong here, but this makes no sense. Carriers
> surcharge like this precisely to guard against call center, predictive and
> other mass outbound calling scenarios.
>
>
>
> It just doesn't make since, math wise, that individuals hanging up on voice
> mail are going to significantly impact overall ACD stats, etc. So unless you
> have a very strange set of use cases or are pushing another category of
> traffic (ie call center) that skews you overall relationship with the
> carrier - I would go back and re-negotiate your arrangement.
>
>
>
> Yes, FS is a b2bua and all is possible. But it is probably a better use of
> time to approach this as a business issue.
>
>
>
> My 2 cents.
>
>
>
>
>
> On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:
>
> How can FS force a Minimum call duration for a FS caller (someone calling
> out of FS)?
>
>
>
> We have a carrier that penalizes us with a surcharge for short duration
> calls (sound familiar?).
>
>
>
> So when a FS caller (not a call center or predictive dialer) calls a cell
> phone and gets a ring tone or calls an answering machine, the FS caller
> hangs up because they do not want to leave a message.  But they do this in
> less then a few seconds after the call is answered.  This becomes a short
> duration call and bang the surcharge applies.  It is actually cheaper to pay
> for a longer call time (6 seconds in this case) and avoid the short duration
> surcharge.  But the FS caller does not know this.
>
>
>
> So, how can FS hold the connection to the called party open for at least the
> minimum amount of time I need to avoid the short call charge? even though my
> FS caller has already hung up the phone on his end?  I would like to do this
> in the xml dialplanif possible.
>
>
>
> Thanks
>
>
>
> -Frank
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gilbertandrew at me.com  Tue Dec  9 16:27:36 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 09 Dec 2008 19:27:36 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <09aa01c95a4e$7b7a6ea0$33014c0a@ws4>
References: <09aa01c95a4e$7b7a6ea0$33014c0a@ws4>
Message-ID: <D149AA89-71BB-4626-918D-9B1845EBEC9B@me.com>

What do your records say? Ie do they balance to what the carrier  
claims? You should at a minimum have macro level data to confirm  
against.

27% seems high, but even at that level if you assume your remaining  
population is "normal" you are still no where close to call center /  
predictive traffic in the overall sense. For example, 2 minutes ACD on  
the normal population is still almost 90 seconds overall. Compare this  
to outbound call centers that might have an overall ACD in the 10-30  
second range and have well over 50%, probably much higher, short  
duration. I would tell your carrier to stop being silly, or find  
another one.

I am unsure you can do it just in the dialplan, but it is a somewhat  
trivial app. The issue is it is difficult to safely avoid scenarios  
where leg B might actually be a real person, talking to dead air. This  
is not good citizenship. It breaks implicit assumptions about network  
behavior and is unfair to end users. It is illegal if applied to a  
predictive scenario.


On Dec 9, 2008, at 5:35 PM, Frank @ Impact wrote:

> On our last bill, the carrier said we had 27% short duration calls  
> (maybe they are wrong but it was on the bill).  It is definitely not  
> call center. But these callers hangup as soon as they hear answer  
> machine or most of the time a ring back tone from cell phone.  This  
> class of caller will call a cell phone, hear the ring back, hangup  
> right away and then call back another 2 minutes later and repeat the  
> cycle.
>
> So, if I have to make it work the way I suggested (hold the  
> connection open for at least the minimum time, how might you suggest  
> I do it in the dial plan?
>
> -----Original Message-----
>
> Don't want the tone to be wrong here, but this makes no sense.  
> Carriers surcharge like this precisely to guard against call center,  
> predictive and other mass outbound calling scenarios.
>
> It just doesn't make since, math wise, that individuals hanging up  
> on voice mail are going to significantly impact overall ACD stats,  
> etc. So unless you have a very strange set of use cases or are  
> pushing another category of traffic (ie call center) that skews you  
> overall relationship with the carrier - I would go back and re- 
> negotiate your arrangement.
>
> Yes, FS is a b2bua and all is possible. But it is probably a better  
> use of time to approach this as a business issue.
>
> My 2 cents.
>
>
> On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:
>
>
> How can FS force a Minimum call duration for a FS caller (someone  
> calling out of FS)?
>
> We have a carrier that penalizes us with a surcharge for short  
> duration calls (sound familiar?).
>
> So when a FS caller (not a call center or predictive dialer) calls a  
> cell phone and gets a ring tone or calls an answering machine, the  
> FS caller hangs up because they do not want to leave a message.  But  
> they do this in less then a few seconds after the call is answered.   
> This becomes a short duration call and bang the surcharge applies.   
> It is actually cheaper to pay for a longer call time (6 seconds in  
> this case) and avoid the short duration surcharge.  But the FS  
> caller does not know this.
>
> So, how can FS hold the connection to the called party open for at  
> least the minimum amount of time I need to avoid the short call  
> charge? even though my FS caller has already hung up the phone on  
> his end?  I would like to do this in the xml dialplanif possible.
>
> Thanks
>
> -Frank
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frank at impactfax.com  Tue Dec  9 17:10:41 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 9 Dec 2008 20:10:41 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <87f2f3b90812091553s561b6b48kda080b1bee0dd775@mail.gmail.com>
Message-ID: <0a4701c95a64$1ff68990$33014c0a@ws4>

Pretty simple...

<?xml version="1.0"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="Regex/XML Dialplan">
    <context name="default">
      <extension name="myextension">
        <condition>
          <action application="export"
data="effective_caller_id_name="/>
          <action application="export"
data="effective_caller_id_number=5555551212"/>
	  <action application="enum" data="5555552121"/>
	  <action application="bridge" data="${enum_auto_route}"/> 
	  <action application="hangup"/>
        </condition>
      </extension>
    </context>
  </section>
</document>

-----Original Message-----
 
Can you paste your dialplan entry here? I have some thoughts but it
would be better if I knew what you were doing before I go any further.
-MC



On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <frank at impactfax.com>
wrote:
> On our last bill, the carrier said we had 27% short duration calls
(maybe
> they are wrong but it was on the bill).  It is definitely not call
center.
> But these callers hangup as soon as they hear answer machine or most
of the
> time a ring back tone from cell phone.  This class of caller will call
a
> cell phone, hear the ring back, hangup right away and then call back
another
> 2 minutes later and repeat the cycle.
>
>
>
> So, if I have to make it work the way I suggested (hold the connection
open
> for at least the minimum time, how might you suggest I do it in the
dial
> plan?
>
>
>
> -----Original Message-----
>
>
> Don't want the tone to be wrong here, but this makes no sense.
Carriers
> surcharge like this precisely to guard against call center, predictive
and
> other mass outbound calling scenarios.
>
>
>
> It just doesn't make since, math wise, that individuals hanging up on
voice
> mail are going to significantly impact overall ACD stats, etc. So
unless you
> have a very strange set of use cases or are pushing another category
of
> traffic (ie call center) that skews you overall relationship with the
> carrier - I would go back and re-negotiate your arrangement.
>
>
>
> Yes, FS is a b2bua and all is possible. But it is probably a better
use of
> time to approach this as a business issue.
>
>
>
> My 2 cents.
>
>
>
>
>
> On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:
>
> How can FS force a Minimum call duration for a FS caller (someone
calling
> out of FS)?
>
>
>
> We have a carrier that penalizes us with a surcharge for short
duration
> calls (sound familiar?).
>
>
>
> So when a FS caller (not a call center or predictive dialer) calls a
cell
> phone and gets a ring tone or calls an answering machine, the FS
caller
> hangs up because they do not want to leave a message.  But they do
this in
> less then a few seconds after the call is answered.  This becomes a
short
> duration call and bang the surcharge applies.  It is actually cheaper
to pay
> for a longer call time (6 seconds in this case) and avoid the short
duration
> surcharge.  But the FS caller does not know this.
>
>
>
> So, how can FS hold the connection to the called party open for at
least the
> minimum amount of time I need to avoid the short call charge. even
though my
> FS caller has already hung up the phone on his end?  I would like to
do this
> in the xml dialplanif possible.
>
>
>
> Thanks
>
>
>
> -Frank
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From ack at telefonica.net  Tue Dec  9 18:10:10 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Wed, 10 Dec 2008 03:10:10 +0100
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <191c3a030812040734s4f514f42s9a30a48c93709fd5@mail.gmail.com>
References: <1228352588.25709.42.camel@develop4>
	<191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
	<1228365981.25709.60.camel@develop4>
	<191c3a030812040734s4f514f42s9a30a48c93709fd5@mail.gmail.com>
Message-ID: <1228875010.2477.67.camel@develop4>

Thanks Anthony , you did a great work ! this is fixed in svn r10691.

Some notes for people using Sonus and L3 as was my case :

in var.xml in some scenario you may need :

<X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/>

in sip_profiles/internal.xml :

<param name="rtp-rewrite-timestamps" value="true"/>

might help for some people with rtp issues :

<param name="rtp-timer-name" value="none"/>

If you have issues with DTMF and timestamps add also :

<param name="pass-rfc2833" value="true"/>

I've a little issues with DTMF from VOIP , i i'll figure out can could
be the issue , from PSTN all works like a charm :)

Cheers,

El jue, 04-12-2008 a las 09:34 -0600, Anthony Minessale escribi?:
> most likely it's because during the time you are dong artificial
> ringback the other side is not doing RTP right.
> 
> When the call is answered we flush the rtp buffer and your missing
> audio is probably flushed with it.
> so you can choose to have a 3 second delay or erase the 3 seconds as
> it does now.
> 
> This is a known problem with sonus which has been proven to build up
> an audio delay during the time
> you are waiting for the call to answer.  I'm sure you prefer the way
> it is to a large audio delay.
> 
> 
> 
> On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <ack at telefonica.net>
> wrote:
>         No TDM , all is SIP :
>         
>         
>         PSTN ---> Sip Proxy_A --> FS ( brigde )
>         ringback/transfer_ringback
>         -> Sip Proxy_B --> PSTN
>         
>         
>         In logfile i think you can get some details about Media
>         Gateways
>         ( Sonus ) PSTN inbound / outbound is provided by Level3.
>         
>         I can get a capture of a call if you want, in capture the
>         audio is not
>         missing, issue with :
>         
>         - rtp buffer ?
>         - Sonus ?
>         
>         Let me know anything you need so i can provide a log or create
>         a new
>         scenario.
>         
>         
>         Thanks,
>         
>         El mi?, 03-12-2008 a las 22:12 -0600, Anthony Minessale
>         escribi?:
>         
>         > what does PSTN represent?
>         >
>         > I know what the PSTN is but how are you reaching it?
>         > is it TDM, SIP etc... what gateway type other details.
>         >
>         >
>         > On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero
>         <ack at telefonica.net>
>         > wrote:
>         >         Hi guys,
>         >
>         >          I've a strange issue with FS , version svn
>         -r10584 ,
>         >         when FS bridges a call first 3 seconds of audio are
>         missing ,
>         >         looks that
>         >         only happens on PSTN calls and using ringback or
>         >         transfer_ringback. This
>         >         only happens in calls from PSTN , not from VOIP.
>         Some
>         >         scenarios i tried
>         >         to isolate this issue :
>         >
>         >
>         >         - Issue
>         >
>         >         PSTN --> FS ( brigde ) ringback/transfer_ringback ->
>         PSTN
>         >
>         >         - Good setting bypass_media before run bridge but i
>         need rtp
>         >         in FS path
>         >
>         >         PSTN --> FS ( brigde ) ringback/transfer_ringback ->
>         PSTN
>         >
>         >         - Good
>         >
>         >         PSTN --> FS ( brigde ) WITHOUT
>         ringback/transfer_ringback ->
>         >         PSTN
>         >
>         >         - Always good
>         >
>         >         VOIP --> FS ( brigde ) -> PSTN
>         >
>         >
>         >         Dialplan has nothing wrong ( i guess ):
>         >
>         >         <extension name="Transfers">
>         >            <condition field="destination_number"
>         >         expression="^1??XXXXXXXXXX$">
>         >              <action application="answer"/>
>         >              <action application="speak" data="cepstral|
>         Allison-8kHz|
>         >         blah"/>
>         >              <action application="set"
>         >         data="hangup_after_bridge=false"/>
>         >              <action application="set"
>         data="playback_terminators=#"/>
>         >              <action application="set" data="ringback=
>         $${us-ring}"/>
>         >              <action application="set"
>         data="transfer_ringback=
>         >         $${hold_music}"/>
>         >              <action application="set"
>         data="effective_caller_id_name=
>         >         ${caller_id_name}"/>
>         >              <action application="set"
>         >         data="effective_caller_id_number=
>         >         ${caller_id_number}"/>
>         >              <action application="set"
>         data="originate_timeout=30"/>
>         >              <action application="set"
>         data="call_timeout=30"/>
>         >              <action application="bridge"
>         >         data="sofia/default/18008226235 at PSTN_GW"/>
>         >              <action application="speak" data="cepstral|
>         Allison-8kHz|
>         >         Transfer
>         >         finished"/>
>         >              <action application="hangup"/>
>         >            </condition>
>         >          </extension>
>         >
>         >
>         >
>         >         Any ideas ?
>         >
>         >         Attached log of FS ( F8 from console ).
>         >
>         >
>         >         Thanks in advance !
>         >
>         >         --
>         >         Angel Carpintero
>         >         ack ( at ) telefonica ( dot ) net
>         >
>         >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF
>         AC2C CA61
>         >         6EF1 B90D
>         >
>         >
>         >
>         
>         > --
>         > Anthony Minessale II
>         >
>         > FreeSWITCH http://www.freeswitch.org/
>         > ClueCon http://www.cluecon.com/
>         >
>         > AIM: anthm
>         > MSN:anthony_minessale at hotmail.com
>         > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>         > IRC: irc.freenode.net #freeswitch
>         >
>         > FreeSWITCH Developer Conference
>         > sip:888 at conference.freeswitch.org
>         > iax:guest at conference.freeswitch.org/888
>         > googletalk:conf+888 at conference.freeswitch.org
>         > pstn:213-799-1400
>         
>         
>         --
>         
>         Angel Carpintero
>         ack ( at ) telefonica ( dot ) net
>         
>         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
>         6EF1 B90D
>         
>         
>         
>         _______________________________________________
>         Freeswitch-users mailing list
>         Freeswitch-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

"No basta saber, hay que aplicar lo que se sabe; 
no basta querer hacerlas cosas, hay que hacerlas".

"Knowing is not enough; we must apply. 
 Willing is not enough; we must do"

 Johann Wolfgang von Goethe
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From jpalley at idapted.com  Tue Dec  9 18:58:38 2008
From: jpalley at idapted.com (Jonathan Palley)
Date: Wed, 10 Dec 2008 10:58:38 +0800
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <B7BC254D-855D-4247-82B9-BEF52C024E9E@jerris.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
	<B7BC254D-855D-4247-82B9-BEF52C024E9E@jerris.com>
Message-ID: <2d8777c00812091858s1c5fa3d9m4e5f11163b894628@mail.gmail.com>

I can offer a bit of a bounty for this.  Can anyone else chip in?

Thanks -
JP

On Tue, Dec 9, 2008 at 11:45 PM, Michael Jerris <mike at jerris.com> wrote:

> It is something we have been discussing as we need these stats to do
> rtcp properly but we have not written any code to do so.  It is
> "somewhat" difficult.  I would say it is on our minds but not on any
> roadmap just yet.
>
> MIke
>
> On Dec 8, 2008, at 11:37 PM, Jonathan Palley wrote:
>
> > I'm curious to start a discussion on being able to query a channel
> > and get statistics on the incoming jitter and packet loss
> > (calculated from the RTP, not RTCP).
> >
> > Is this on the roadmap?  Is it hard to do?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jonathan Palley | Idapted Inc.
jpalley at idapted.com
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From c_cav_01 at yahoo.com  Tue Dec  9 20:24:29 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 9 Dec 2008 20:24:29 -0800 (PST)
Subject: [Freeswitch-users]  incoming call routing <domain>
Message-ID: <20928933.post@talk.nabble.com>


Cable modem <----> nat router <---->  fs

fs is set as DMZ on nat router so all packets get there.

My ipv4 address is 192.168.0.x  The nat router holds the public IP.  Public
IP is a registered domain sparkz.tv so addressable from the internet cloud. 
Since fs is DMZ, all requests for sparkz.tv or sip.sparkz.tv are resolved
and so IP routing is good.

So I'm trying to get external sip/soft phones registered and routed
properly.  The domain/server set in the phone client is sip.sparkz.tv:5080,
since the wiki says they need to be set that way???

I have created a conf/directory/sip.sparkz.tv.xml  and a
conf/directory/sip.sparkz.tv where I have users registration info.

conf/directory/sip.sparkz.tv.xml was copied from default.xml and has:
      
    </params>


I have modified conf/sip_profiles/external.xml and added an <alias
name="sip.sparkz.tv"/>

External phones are registering and are visible under sofia status profiles
external and sip.sparkz.tv

Calls outbound from the phones are being routed properly.

Calls inbound to their DID's are not.
Calls to softphones on the local private net 192.168.0.x register and route
properly.
vars.xml sets domain to ip_v4...
the default.xml dialplan seems to iif the profile to either nat or default.. 
so I end up with the call going to DID at 192.168.0.x rather than the
registered interface...

I'm routing the calls in the dialplan to bridge to user/$1@$${domain}  but
$${domain} is set to ip_v4 so it's wrong...

Any clues what I need to do next to get them routing properly?  I want to be
able to support multiple domains. how do I do this correctly?
-- 
View this message in context: http://www.nabble.com/incoming-call-routing-%3Cdomain%3E-tp20928933p20928933.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From c_cav_01 at yahoo.com  Tue Dec  9 20:25:32 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 9 Dec 2008 20:25:32 -0800 (PST)
Subject: [Freeswitch-users]  incoming call routing <domain>
Message-ID: <20928933.post@talk.nabble.com>


Cable modem <----> nat router <---->  fs

fs is set as DMZ on nat router so all packets get there.

My ipv4 address is 192.168.0.x  The nat router holds the public IP.  Public
IP is a registered domain sparkz.tv so addressable from the internet cloud. 
Since fs is DMZ, all requests for sparkz.tv or sip.sparkz.tv are resolved
and so IP routing is good.

So I'm trying to get external sip/soft phones registered and routed
properly.  The domain/server set in the phone client is sip.sparkz.tv:5080,
since the wiki says they need to be set that way???

I have created a conf/directory/sip.sparkz.tv.xml  and a
conf/directory/sip.sparkz.tv where I have users registration info.

conf/directory/sip.sparkz.tv.xml was copied from default.xml and has:

      param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/${dialed_user}@${dialed_domain})}"
    /params


I have modified conf/sip_profiles/external.xml and added an <alias
name="sip.sparkz.tv"/>

External phones are registering and are visible under sofia status profiles
external and sip.sparkz.tv

Calls outbound from the phones are being routed properly.

Calls inbound to their DID's are not.
Calls to softphones on the local private net 192.168.0.x register and route
properly.
vars.xml sets domain to ip_v4...
the default.xml dialplan seems to iif the profile to either nat or default.. 
so I end up with the call going to DID at 192.168.0.x rather than the
registered interface...

I'm routing the calls in the dialplan to bridge to user/$1@$${domain}  but
$${domain} is set to ip_v4 so it's wrong...

Any clues what I need to do next to get them routing properly?  I want to be
able to support multiple domains. how do I do this correctly?
-- 
View this message in context: http://www.nabble.com/incoming-call-routing-%3Cdomain%3E-tp20928933p20928933.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From dave at 3c.co.uk  Tue Dec  9 21:26:27 2008
From: dave at 3c.co.uk (David Knell)
Date: Wed, 10 Dec 2008 05:26:27 +0000
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493ED9E6.3000803@junctionnetworks.com>
References: <493ED9E6.3000803@junctionnetworks.com>
Message-ID: <493F5303.30208@3c.co.uk>

Hi Erick,

Not sure if you've tried this (or if it'll help), but you can force 
routing in the dialplan like so:
<action application="set" data="sip_h_Route=<sip:@11.22.33.44;lr>" />
<action application="bridge" data="sofia/gateway/gw/$1"/>

Cheers --

Dave


> i forgot to give you the pastebin URL
> http://pastebin.freeswitch.org/6379
>
>   
>> I'm running latest trunk - Revision: 10682
>>
>> I've been doing an ngrep on my external freeswitch SIP port and FS
>> is not sending any SIP packets anywhere when I run the following command.
>> Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are 
>> logged.
>>
>> originate
>> 'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' 
>> &echo()
>>
>> Also, just to be clear, when I remove ";fs_path=..." from the command
>> above a call
>> is set up normally to erick at ejjohnson.org and the SIP packets are 
>> logged
>> to console.
>>
>> Thanks guys.
>>
>>     
>>> What SVN rev are you running? Also, could you do a SIP trace?
>>> TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
>>> Pastebin the output of that and we'll take it from there.
>>> -MC
>>>
>>> On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
>>> <erick at junctionnetworks.com> wrote:
>>>       
>>>> Both:
>>>>
>>>> originate sofia/external/'erick at
>>>>         
>>> ejjohnson.org;fs_path=proxybeta.foo.net'
>>>       
>>>> &echo()
>>>> originate sofia/external/erick at
>>>>         
>>> ejjohnson.org;fs_path=proxybeta.foo.net
>>>       
>>>> &echo()
>>>>
>>>> produce the exact same result & log
>>>>
>>>> :(
>>>>
>>>>         
>>>>> * I think you need to '' the sofia uri /b
>>>>>           
>>>>         
>>     
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From helmut.kuper at ewetel.de  Tue Dec  9 23:35:36 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 10 Dec 2008 08:35:36 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>
	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>
Message-ID: <493F7148.40705@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

on my ubuntu 8.04 I have libtiff4 and libtiff4-dev installed. libtiff
and libtiff-dev is not installed. I gonna test it today

regards
helmut


Michael Collins schrieb:
> Helmut,
> I think Mike J was pointing out that spandsp needs libtiff and
> libtiff-devel in order to compile, so you need to do that first and
> then compile freeswitch.
> -MC
> 
> On Tue, Dec 9, 2008 at 8:09 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> Hi Michael,
> 
> don't know if you get me right: Everything is there, but obviously FS
> makefile has to compile "libs/spandsp/src" before mod_fax (at least I
> guess so). Currently the Makefile referred to libspandsp.la before it is
> compiled.
> 
> regards
> helmut
> 
> 
> 
> Michael Jerris schrieb:
>>>> make sure you have libtiff and libtiff dev packages installed then re-
>>>> configure freeswitch
>>>>
>>>> Mike
>>>>
>>>> On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:
>>>>
>>>> Hello,
>>>>
>>>> I tried to compile mod_fax today with trunk from yesterday. A 'make'
>>>> in
>>>> FS trunk directory led to an error saying that libspandsp.la wasn't
>>>> found in libs/spandsp/src. So I had to configure and compile (make)
>>>> spandsp manually before compiling FS.
>>>>
>>>> regards
>>>> helmut
>>>>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
> 
>>
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
>>

> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From jalsot at gmail.com  Wed Dec 10 00:23:26 2008
From: jalsot at gmail.com (Tamas)
Date: Wed, 10 Dec 2008 09:23:26 +0100
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <2d8777c00812091858s1c5fa3d9m4e5f11163b894628@mail.gmail.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>	<B7BC254D-855D-4247-82B9-BEF52C024E9E@jerris.com>
	<2d8777c00812091858s1c5fa3d9m4e5f11163b894628@mail.gmail.com>
Message-ID: <493F7C7E.6030805@gmail.com>

Hello,

I've added bounty for RTCP already:
http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support

I know that the requester wants stats for RTP but maybe we could make a
joint bounty ;)

Regards,
    Tamas

Jonathan Palley ?rta:
> I can offer a bit of a bounty for this.  Can anyone else chip in?
>
> Thanks -
> JP
>
> On Tue, Dec 9, 2008 at 11:45 PM, Michael Jerris <mike at jerris.com
> <mailto:mike at jerris.com>> wrote:
>
>     It is something we have been discussing as we need these stats to do
>     rtcp properly but we have not written any code to do so.  It is
>     "somewhat" difficult.  I would say it is on our minds but not on any
>     roadmap just yet.
>
>     MIke
>
>     On Dec 8, 2008, at 11:37 PM, Jonathan Palley wrote:
>
>     > I'm curious to start a discussion on being able to query a channel
>     > and get statistics on the incoming jitter and packet loss
>     > (calculated from the RTP, not RTCP).
>     >
>     > Is this on the roadmap?  Is it hard to do?
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Jonathan Palley | Idapted Inc.
> jpalley at idapted.com <mailto:jpalley at idapted.com>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Wed Dec 10 01:45:34 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 03:45:34 -0600
Subject: [Freeswitch-users] incoming call routing <domain>
In-Reply-To: <20928933.post@talk.nabble.com>
References: <20928933.post@talk.nabble.com>
Message-ID: <498842E3-2204-4204-AF40-43AA344E0C35@freeswitch.org>

Join IRC so you can interact with people real time.  Your setup  
require a deep understanding of SIP and FreeSWITCH to setup correctly.

/b

On Dec 9, 2008, at 10:25 PM, ccav wrote:

>
> Cable modem <----> nat router <---->  fs
>
> fs is set as DMZ on nat router so all packets get there.
>
> My ipv4 address is 192.168.0.x  The nat router holds the public IP.   
> Public
> IP is a registered domain sparkz.tv so addressable from the internet  
> cloud.
> Since fs is DMZ, all requests for sparkz.tv or sip.sparkz.tv are  
> resolved
> and so IP routing is good.
>
> So I'm trying to get external sip/soft phones registered and routed
> properly.  The domain/server set in the phone client is  
> sip.sparkz.tv:5080,
> since the wiki says they need to be set that way???
>
> I have created a conf/directory/sip.sparkz.tv.xml  and a
> conf/directory/sip.sparkz.tv where I have users registration info.
>
> conf/directory/sip.sparkz.tv.xml was copied from default.xml and has:
>
>      param name="dial-string"
> value="{presence_id=${dialed_user}@$ 
> {dialed_domain},transfer_fallback_extension=${dialed_user}}$ 
> {sofia_contact(${dialed_domain}/${dialed_user}@${dialed_domain})}"
>    /params
>
>
> I have modified conf/sip_profiles/external.xml and added an <alias
> name="sip.sparkz.tv"/>
>
> External phones are registering and are visible under sofia status  
> profiles
> external and sip.sparkz.tv
>
> Calls outbound from the phones are being routed properly.
>
> Calls inbound to their DID's are not.
> Calls to softphones on the local private net 192.168.0.x register  
> and route
> properly.
> vars.xml sets domain to ip_v4...
> the default.xml dialplan seems to iif the profile to either nat or  
> default..
> so I end up with the call going to DID at 192.168.0.x rather than the
> registered interface...
>
> I'm routing the calls in the dialplan to bridge to user/$1@$$ 
> {domain}  but
> $${domain} is set to ip_v4 so it's wrong...
>
> Any clues what I need to do next to get them routing properly?  I  
> want to be
> able to support multiple domains. how do I do this correctly?
> -- 
> View this message in context: http://www.nabble.com/incoming-call-routing-%3Cdomain%3E-tp20928933p20928933.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From carole.olivier at enst.fr  Wed Dec 10 05:37:05 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 10 Dec 2008 05:37:05 -0800 (PST)
Subject: [Freeswitch-users]  segmentation fault by auto record
Message-ID: <20935513.post@talk.nabble.com>


Hello,

I have intalled Freeswitch from opensuse.org as a rpm. I have opensuse 10.3.
I did not make any big configuration, I have just changed a little the
default dialplan and adapted some other files like conference.conf.xml.

I have created a new profile in conference.conf.xml and add the command to
order the automatic record of the conferences:

<profile name="profilespeaker">
   
   
   
   
   
   
   
   
   
</profile>

I have the following extension in my dialplan that uses this profile:

<extension name="call_speakers">
 <condition field="destination_number" expression="^0911$">
   <action application="set"
data="conference_auto_outcall_caller_id_name=call_speakers" /> 
   <action application="set" data="conference_auto_outcall_id_number=0911"
/> 
   <action application="set" data="conference_auto_outcall_timeout=60" /> 
   <action application="set" data="conference_auto_outcall_flags=mute" /> 
   <action application="set" data="api_hangup_hook=conference conf_speaker
kick all" /> 
<!--  devices called  --> 
  <action application="conference_set_auto_outcall"
data="user/1021@$${domain}" /> 
  <action application="conference_set_auto_outcall"
data="user/1022@$${domain}" /> 
<!-- creation conference --> 
  <action application="conference" data="conf_speaker at profilespeaker" /> 
 </condition>
</extension>

1021 and 1022 are IP loud speakers.
I call them with 0911, they answer, I can talk and everything works well.
However, when I hang up a segmentation fault appears and freeswitch
shutdowns. In recordings I can find a file which corresponds to the recorded
call but this is empty. Nothing has been recorded.
If I comment the line with the recording command then it works without
problem except the recording...

I have joined two files: one contains the errors that appears when I run
freeswitch and the other what happens if I call the extension 0911.

http://www.nabble.com/file/p20935513/running_freeswitch.txt
running_freeswitch.txt 
http://www.nabble.com/file/p20935513/call_extension.txt call_extension.txt 

If someone has an idea, it would be very helpful
Thanks!!
Carole
-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20935513.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From carole.olivier at enst.fr  Wed Dec 10 05:50:09 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 10 Dec 2008 05:50:09 -0800 (PST)
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20935513.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
Message-ID: <20935752.post@talk.nabble.com>


Sorry, here is the profile profile speaker

<profile name="profilespeaker">
  < param name="rate" value="8000" /> 
  < param name="interval" value="20" /> 
  < param name="energy-level" value="300" /> 
  < param name="caller-id-name" value="$${outbound_caller_name}" /> 
  < param name="caller-id-number" value="$${outbound_caller_id}" /> 
  < param name="comfort-noise-level" value="1400" /> 
  < param name="comfort-noise" value="true" /> 
  < param name="member_flags" value="waste" /> 
  < param name="auto-record"
value="$${base_dir}/recordings/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"
/> 
  </profile>

-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20935752.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From john at loopfx.com  Wed Dec 10 06:36:33 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 09:36:33 -0500
Subject: [Freeswitch-users] No audio after transfer
Message-ID: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>

Sorry to repost, but I haven't heard anything back on this in a little
while.

 

I checked out the trunk last week.  I'm on revision 10597.  

 

Thanks,

John

 

From: John Rutherford 
Sent: Monday, December 08, 2008 4:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: No audio after transfer

 

I'm trying to get an attended transfer work with freeSWITCH, but it's
not quite working.  I have Microsoft Speech Server on one side and
Televantage on the other.  

 

MSS is originating a call, which freeSWITCH is bridging to Televantage.
That calls connects just fine.  Then, MSS sends a re-INVITE to
Televantage to put the call on hold.  This works too.  Then, MSS
originates another call to freeSWITCH, which is again bridged to
Televantage.  This works fine too.  

 

Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
should be complete, but there is no audio between the two calls-just
silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not
seeing anything obviously wrong.  

 

After the REFER, I can see audio for both calls going between freeSWITCH
and Televantage, so it seems that the only thing missing is freeSWITCH
routing the audio from one call to the other call and vice-versa.

 

 

Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this.  

 

Thanks!

John

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From frank at impactfax.com  Wed Dec 10 06:46:01 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Wed, 10 Dec 2008 09:46:01 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <87f2f3b90812091553s561b6b48kda080b1bee0dd775@mail.gmail.com>
Message-ID: <0cd801c95ad6$055cf890$33014c0a@ws4>

This is a bit beyond me, but in the xml dialplan could we use the
execute_on_answer to transfer to an extension that sets up an intercept
of the bleg on hangup_after_bridge?  Or use the api_hangup_hook to
transfer the bleg to another extension after the aleg hangs up?

I have been reading all the wiki information I can and these smell like
they might help.  But it is not clear to me how they would be pieced
together exactly to achieve this end.

MC, any thoughts?

-----Original Message-----
 
Can you paste your dialplan entry here? I have some thoughts but it
would be better if I knew what you were doing before I go any further.
-MC



On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <frank at impactfax.com>
wrote:
> On our last bill, the carrier said we had 27% short duration calls
(maybe
> they are wrong but it was on the bill).  It is definitely not call
center.
> But these callers hangup as soon as they hear answer machine or most
of the
> time a ring back tone from cell phone.  This class of caller will call
a
> cell phone, hear the ring back, hangup right away and then call back
another
> 2 minutes later and repeat the cycle.
>
>




From anthony.minessale at gmail.com  Wed Dec 10 07:34:27 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Dec 2008 09:34:27 -0600
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
Message-ID: <191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>

Are you expecting every message that MSS sends FS to be in turn sent to
televantage?
That is a proxy behaviour.  Since FreeSWITCH plays the role of a b2bua it
will not pass the
messages across a bridge.



On Wed, Dec 10, 2008 at 8:36 AM, John Rutherford <john at loopfx.com> wrote:

>  Sorry to repost, but I haven't heard anything back on this in a little
> while.
>
>
>
> I checked out the trunk last week.  I'm on revision 10597.
>
>
>
> Thanks,
>
> John
>
>
>
> *From:* John Rutherford
> *Sent:* Monday, December 08, 2008 4:36 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* No audio after transfer
>
>
>
> I'm trying to get an attended transfer work with freeSWITCH, but it's not
> quite working.  I have Microsoft Speech Server on one side and Televantage
> on the other.
>
>
>
> MSS is originating a call, which freeSWITCH is bridging to Televantage.
> That calls connects just fine.  Then, MSS sends a re-INVITE to Televantage
> to put the call on hold.  This works too.  Then, MSS originates another call
> to freeSWITCH, which is again bridged to Televantage.  This works fine too.
>
>
>
>
> Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
> should be complete, but there is no audio between the two calls?just
> silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not seeing
> anything obviously wrong.
>
>
>
> After the REFER, I can see audio for both calls going between freeSWITCH
> and Televantage, so it seems that the only thing missing is freeSWITCH
> routing the audio from one call to the other call and vice-versa.
>
>
>
>
>
> Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
> logs, and I can easily reproduce this.
>
>
>
> Thanks!
>
> John
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Wed Dec 10 07:41:52 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 07:41:52 -0800
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <0cd801c95ad6$055cf890$33014c0a@ws4>
References: <87f2f3b90812091553s561b6b48kda080b1bee0dd775@mail.gmail.com>
	<0cd801c95ad6$055cf890$33014c0a@ws4>
Message-ID: <87f2f3b90812100741r2067549fmc88674cf99af7b7e@mail.gmail.com>

On Wed, Dec 10, 2008 at 6:46 AM, Frank @ Impact <frank at impactfax.com> wrote:
> This is a bit beyond me, but in the xml dialplan could we use the
> execute_on_answer to transfer to an extension that sets up an intercept
> of the bleg on hangup_after_bridge?  Or use the api_hangup_hook to
> transfer the bleg to another extension after the aleg hangs up?
>
> I have been reading all the wiki information I can and these smell like
> they might help.  But it is not clear to me how they would be pieced
> together exactly to achieve this end.
>
> MC, any thoughts?
>

I spent a fair amount of time playing with these last night but I
didn't find a solution. I'm still thinking about it, but I believe
that most likely it will require some scripting in Lua (or another
language) to be able to get this to work. This is the first time I've
ever dealt with keeping the b-leg alive when the a-leg hangs up - it
is usually the other way around.

I recommend you read up on Lua scripting because you're gonna need it
to be able to pull this off. (You could use a different language but
Lua is the scripting language of choice amongst the FS devs so I
highly recommend using it.)

I will tinker with this a bit later today when I have some more time.
In the meantime if you could start experimenting with the ideas you've
presented and brush up on Lua that would be great. BTW, are you on
IRC?

Thanks,
MC

> -----Original Message-----
>
> Can you paste your dialplan entry here? I have some thoughts but it
> would be better if I knew what you were doing before I go any further.
> -MC
>
>
>
> On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <frank at impactfax.com>
> wrote:
>> On our last bill, the carrier said we had 27% short duration calls
> (maybe
>> they are wrong but it was on the bill).  It is definitely not call
> center.
>> But these callers hangup as soon as they hear answer machine or most
> of the
>> time a ring back tone from cell phone.  This class of caller will call
> a
>> cell phone, hear the ring back, hangup right away and then call back
> another
>> 2 minutes later and repeat the cycle.
>>
>>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Wed Dec 10 07:53:09 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 07:53:09 -0800
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20935513.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
Message-ID: <87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>

Thanks for reporting this. It would be helpful to know a bit more. Can
you start freeswitch and press F12 (or type "version" at the CLI) and
report back what it says?
Also, a backtrace (bt) is generally useful. If you could produce a
"bt" and a "bt full" from you core file that would be extremely
helpful.

see this link for more information:
http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

you should have a "core" file for each segfault that occurred. Use the
gdb program to get the back trace:

gdb /path/to/fs/binary core.xxx

then capture the output from these two commands:

bt <enter>
bt full <enter>

When you type those commands you'll see tons of debugging info;
capture that and put it in a pastebin (pastebin.freeswitch.org) then
report back here.

You can exit the gdb debugger by typing q <enter>

Thanks for helping us collect information!

-MC


On Wed, Dec 10, 2008 at 5:37 AM, Carole O. <carole.olivier at enst.fr> wrote:
>
> Hello,
>
> I have intalled Freeswitch from opensuse.org as a rpm. I have opensuse 10.3.
> I did not make any big configuration, I have just changed a little the
> default dialplan and adapted some other files like conference.conf.xml.
>
> I have created a new profile in conference.conf.xml and add the command to
> order the automatic record of the conferences:
>
> <profile name="profilespeaker">
>
>
>
>
>
>
>
>
>
> </profile>
>
> I have the following extension in my dialplan that uses this profile:
>
> <extension name="call_speakers">
>  <condition field="destination_number" expression="^0911$">
>   <action application="set"
> data="conference_auto_outcall_caller_id_name=call_speakers" />
>   <action application="set" data="conference_auto_outcall_id_number=0911"
> />
>   <action application="set" data="conference_auto_outcall_timeout=60" />
>   <action application="set" data="conference_auto_outcall_flags=mute" />
>   <action application="set" data="api_hangup_hook=conference conf_speaker
> kick all" />
> <!--  devices called  -->
>  <action application="conference_set_auto_outcall"
> data="user/1021@$${domain}" />
>  <action application="conference_set_auto_outcall"
> data="user/1022@$${domain}" />
> <!-- creation conference -->
>  <action application="conference" data="conf_speaker at profilespeaker" />
>  </condition>
> </extension>
>
> 1021 and 1022 are IP loud speakers.
> I call them with 0911, they answer, I can talk and everything works well.
> However, when I hang up a segmentation fault appears and freeswitch
> shutdowns. In recordings I can find a file which corresponds to the recorded
> call but this is empty. Nothing has been recorded.
> If I comment the line with the recording command then it works without
> problem except the recording...
>
> I have joined two files: one contains the errors that appears when I run
> freeswitch and the other what happens if I call the extension 0911.
>
> http://www.nabble.com/file/p20935513/running_freeswitch.txt
> running_freeswitch.txt
> http://www.nabble.com/file/p20935513/call_extension.txt call_extension.txt
>
> If someone has an idea, it would be very helpful
> Thanks!!
> Carole
> --
> View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20935513.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Wed Dec 10 07:58:34 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 09:58:34 -0600
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
Message-ID: <8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>

I have already labbed this up on SVN trunk and I don't get a segfault  
but I get something else that prevents it from working properly.    We  
are working on it today.  Also what version are you running?

/b

On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:

> Thanks for reporting this. It would be helpful to know a bit more. Can
> you start freeswitch and press F12 (or type "version" at the CLI) and
> report back what it says?
> Also, a backtrace (bt) is generally useful. If you could produce a
> "bt" and a "bt full" from you core file that would be extremely
> helpful.
>
> see this link for more information:
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>
> you should have a "core" file for each segfault that occurred. Use the
> gdb program to get the back trace:
>
> gdb /path/to/fs/binary core.xxx
>
> then capture the output from these two commands:
>
> bt <enter>
> bt full <enter>
>
> When you type those commands you'll see tons of debugging info;
> capture that and put it in a pastebin (pastebin.freeswitch.org) then
> report back here.
>
> You can exit the gdb debugger by typing q <enter>
>
> Thanks for helping us collect information!
>
> -MC

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From john at loopfx.com  Wed Dec 10 09:51:59 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 12:51:59 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
	<191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>

No. I realize that's it's a B2BUA and that's exactly what we want.  

 

Everything with the transfer seems to work fine, except that there is no
audio.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, December 10, 2008 10:34 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

Are you expecting every message that MSS sends FS to be in turn sent to
televantage?
That is a proxy behaviour.  Since FreeSWITCH plays the role of a b2bua
it will not pass the
messages across a bridge.




On Wed, Dec 10, 2008 at 8:36 AM, John Rutherford <john at loopfx.com>
wrote:

Sorry to repost, but I haven't heard anything back on this in a little
while.

 

I checked out the trunk last week.  I'm on revision 10597.  

 

Thanks,

John

 

From: John Rutherford 
Sent: Monday, December 08, 2008 4:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: No audio after transfer

 

I'm trying to get an attended transfer work with freeSWITCH, but it's
not quite working.  I have Microsoft Speech Server on one side and
Televantage on the other.  

 

MSS is originating a call, which freeSWITCH is bridging to Televantage.
That calls connects just fine.  Then, MSS sends a re-INVITE to
Televantage to put the call on hold.  This works too.  Then, MSS
originates another call to freeSWITCH, which is again bridged to
Televantage.  This works fine too.  

 

Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
should be complete, but there is no audio between the two calls-just
silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not
seeing anything obviously wrong.  

 

After the REFER, I can see audio for both calls going between freeSWITCH
and Televantage, so it seems that the only thing missing is freeSWITCH
routing the audio from one call to the other call and vice-versa.

 

 

Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this.  

 

Thanks!

John


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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From brian at freeswitch.org  Wed Dec 10 09:58:46 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 11:58:46 -0600
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
	<191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
Message-ID: <BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org>

would be most helpful to capture a pcap of the entire thing by itself  
start to finish.

/b

On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:

> No. I realize that?s it?s a B2BUA and that?s exactly what we want.
>
> Everything with the transfer seems to work fine, except that there  
> is no audio.

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From john at loopfx.com  Wed Dec 10 10:01:56 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 13:01:56 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
	<BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>

I have a pcap, but I'm not able to see anything obviously wrong with it.


 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Wednesday, December 10, 2008 12:59 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

would be most helpful to capture a pcap of the entire thing by itself
start to finish. 

 

/b

 

On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:





No. I realize that's it's a B2BUA and that's exactly what we want. 

 

Everything with the transfer seems to work fine, except that there is no
audio.

 

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From msc at freeswitch.org  Wed Dec 10 10:08:32 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 10:08:32 -0800
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
	<191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
	<BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org>
	<81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>
Message-ID: <87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com>

On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com> wrote:
> I have a pcap, but I'm not able to see anything obviously wrong with it.

We find that some equipment (in fact a lot of equipment) have features
that cause issues to be quite non-obvious, so perhaps you could give
the pcap to Brian for him to review. He's a total ace when it comes to
bug hunting.

-MC

>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Wednesday, December 10, 2008 12:59 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> would be most helpful to capture a pcap of the entire thing by itself start
> to finish.
>
>
>
> /b
>
>
>
> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>
> No. I realize that's it's a B2BUA and that's exactly what we want.
>
>
>
> Everything with the transfer seems to work fine, except that there is no
> audio.
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mszlazak at aol.com  Wed Dec 10 10:39:42 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 13:39:42 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
Message-ID: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>




I?m trying
to route calls from X-lite <--> FS (Nov. 6 2008 svn) <--> Voceo (Prophecy) to use Voxeo?s ASR instead of FS?s built in PocketSphinx/ASR.



All these applications
reside on the same computer/OS (Win XP). I have at Netgear wifi router that
connects a laptop to my desktop in case that matters but I?m not using the
laptop for any of this.



??



I?ve set up
an extension to bridge calls to Voxeo. Here is the entry in file
conf\dialplan\default.xml:



?





??? <extension name="Doctors
Office">



????? <condition
field="destination_number" expression="^2007$">



??????? <action
application="info"/>



??????? <action application="bridge"
data="sofia/internal/sip:Doctorsoffice at 127.0.0.1:5068"/>??????? 



????? </condition>



??? </extension>



?



I hear one
ring then a hang up. No errors in the FS console and Voxeo?s logs ramp up when
I dial the 2007 extension on X-lite but I do not get any audio from the
dialogue script in Voxeo?s Prophecy ASR called ?Doctorsoffice.? 





Any ideas?



?



Thanks.





Mark.




?


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From brian at freeswitch.org  Wed Dec 10 10:45:22 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 12:45:22 -0600
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>
Message-ID: <4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org>

You're a month behind.. I highly recommend you update.  Chances are  
this has already been fixed.


/b

On Dec 10, 2008, at 12:39 PM, mszlazak at aol.com wrote:

> I?m trying to route calls from X-lite <--> FS (Nov. 6 2008 svn) <-->  
> Voceo (Prophecy) to use Voxeo?s ASR instead of FS?s built in  
> PocketSphinx/ASR.
> All these applications reside on the same computer/OS (Win XP). I  
> have at Netgear wifi=2 0router that connects a laptop to my desktop  
> in case that matters but I?m not using the laptop for any of this.
>
> I?ve set up an extension to bridge calls to Voxeo. Here is the entry  
> in file conf\dialplan\default.xml:
>
>     <extension name="Doctors Office">
>       <condition field="destination_number" expression="^2007$">
>         <action application="info"/>
>         <action application="bridge" data="sofia/internal/sip:Doctorsoffice at 127.0.0.1:5068 
> "/>
>       </condition>
>     </extension>
>
> I hear one ring then a hang up. No errors in the FS console and  
> Voxeo?s logs ramp up when I dial the 2007 extension on X-lite but I  
> do not get any audio from the dialogue script in Voxeo?s Prophecy  
> ASR called ?Doctorsoffice.?
>
> Any ideas?
>
> Thanks.
>
> Mark.
>
> 0A
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mszlazak at aol.com  Wed Dec 10 10:55:21 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 13:55:21 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>
	<4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org>
Message-ID: <8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>


 Yes but someone else I'm in contact with set up FS a couple days ago and is having the same problems. 
Brian should I still update today?


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 10:45 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo









You're a month behind.. I highly recommend you update. ?Chances are this has already been fixed.





/b




On Dec 10, 2008, at 12:39 PM, mszlazak at aol.com wrote:




I?m trying to route calls from X-lite?<-->?FS (Nov. 6 2008?svn)?<-->?Voceo (Prophecy)?to use Voxeo?s ASR instead of FS?s built in PocketSphinx/ASR.

All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi=2 0router that connects a laptop to my desktop in case that matters but I?m not using the laptop for any of this.

??

I?ve set up an extension to bridge calls to Voxeo. Here is the entry?in file conf\dialplan\default.xml:

?

????<extension name="Doctors Office">

??????<condition field="destination_number" expression="^2007$">

????????<action application="info"/>

????????<action application="bridge" data="sofia/internal/sip:Doctorsoffice at 127.0.0.1:5068"/>???????

??????</condition>

????</extension>

?

I20hear one ring then a hang up. No errors in the FS console and Voxeo?s logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeo?s Prophecy ASR called ?Doctorsoffice.??





Any ideas?

?

Thanks.





Mark.


?
0A



Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse.?Start Listening Now!
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=


 





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From brian at freeswitch.org  Wed Dec 10 11:02:49 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 13:02:49 -0600
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>
	<4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org>
	<8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>
Message-ID: <74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org>

If you're not trying this on the latest code then yes I would update  
if possible.  Do you recall who you were talking too?

/b

On Dec 10, 2008, at 12:55 PM, mszlazak at aol.com wrote:

> Yes but someone else I'm in contact with set up FS a couple days ago  
> and is having the same problems.
> Brian should I still update today?
>

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From john at loopfx.com  Wed Dec 10 11:16:07 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 14:16:07 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>
	<87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>

I just emailed it to him.

Thanks!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Wednesday, December 10, 2008 1:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
wrote:
> I have a pcap, but I'm not able to see anything obviously wrong with
it.

We find that some equipment (in fact a lot of equipment) have features
that cause issues to be quite non-obvious, so perhaps you could give
the pcap to Brian for him to review. He's a total ace when it comes to
bug hunting.

-MC

>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian
> West
> Sent: Wednesday, December 10, 2008 12:59 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> would be most helpful to capture a pcap of the entire thing by itself
start
> to finish.
>
>
>
> /b
>
>
>
> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>
> No. I realize that's it's a B2BUA and that's exactly what we want.
>
>
>
> Everything with the transfer seems to work fine, except that there is
no
> audio.
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From mszlazak at aol.com  Wed Dec 10 11:51:30 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 14:51:30 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>
	<74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org>
Message-ID: <8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>


 It was someone from Voxeo support. I think John was the main person helping me with this.

I updated but things got worse all over.
I now can't run other extensions Gino's pizza or some db stuff

Got these errors:

2008-12-10 11:40:23 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll
**The specified module could not be found.? **
2008-12-10 11:40:23 [CONSOLE] mod_spidermonkey.c:944 sm_load_file() Successfully Loaded [C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_core_db.dll]
2008-12-10 11:40:23 [CONSOLE] mod_spidermonkey.c:944 sm_load_file() Successfully Loaded [C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_socket.dll]
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_spidermonkey]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'javascript'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'jsrun'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'jsapi'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_lua]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'lua'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'luarun'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'lua'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_say_en]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:371 switch_loadable_module_process() Adding Say interface 'en'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:118 switch_loadable_module_runtime() Starting runtime thread for CORE_SOFTTIMER_MODULE
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:118 switch_loadable_module_runtime() Starting runtime thread for mod_event_socket
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list dl-candidates default (allow)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 10.0.0.0/8 (deny) to list dl-candidates
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 172.16.0.0/12 (deny) to list dl-candidates
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.0.0/16 (deny) to list dl-candidates
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list rfc1918 default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 10.0.0.0/8 (allow) to list rfc1918
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 172.16.0.0/12 (allow) to list rfc1918
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.0.0/16 (allow) to list rfc1918
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list lan default (allow)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.42.0/24 (deny) to list lan
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.42.42/32 (allow) to list lan
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list strict default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 208.102.123.124/32 (allow) to list strict
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list domains default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:907 switch_load_network_lists() Adding 1.2.3.4/24 (allow) [brian at 10.0.0.2] to list domains
2008-12-10 11:40:23 [CONSOLE] switch_core.c:1258 switch_core_init_and_modload()
FreeSWITCH Version 1.0.trunk (10171M) Started.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]

freeswitch at mark-e40edd7b44> 2008-12-10 11:40:24 [ERR] sofia.c:543 sofia_profile_thread_run() Error Creating SIP UA for profile: internal-ipv6
2008-12-10 11:40:45 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/1000 at 10.0.0.2 [88d1aaf9-a625-444e-883d-c5ac6eeac30e]
2008-12-10 11:40:45 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch->2006 in context default
2008-12-10 11:40:45 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000 at 10.0.0.2] has been answered
2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
2008-12-10 11:40:48 [NOTICE] switch_ivr_async.c:1845 switch_ivr_detect_speech() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2008-12-10 11:40:48 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2) Ended
2008-12-10 11:40:48 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]
2008-12-10 11:41:23 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/1000 at 10.0.0.2 [67882be4-4c10-d248-8fe1-ac02b9b8fc5c]
2008-12-10 11:41:23 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch->8337 in context default
2008-12-10 11:41:23 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000 at 10.0.0.2] has been answered
2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!
2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine!
2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000 at 10.0.0.2) Ended
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]



 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 11:02 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo









If you're not trying this on the latest code then yes I would update if possible. ?Do you recall who you were talking too?



/b




On Dec 10, 2008, at 12:55 PM, mszlazak at aol.com wrote:



Yes but someone else I'm in contact with set up FS a couple days ago and is having the same problems.?
Brian should I still update today?






=


 





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Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Wed Dec 10 11:59:02 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 13:59:02 -0600
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>
	<74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org>
	<8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>
Message-ID: <D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org>

Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.

/b

On Dec 10, 2008, at 1:51 PM, mszlazak at aol.com wrote:

> It was someone from Voxeo support. I think John was the main person  
> helping me with this.
>
> I updated but things got worse all over.
> I now can't run other extensions Gino's pizza or some db stuff
>
>
> 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57  
> switch_core_asr_open() Invalid ASR module [pocketsphinx]!
>  2008-12-10 11:41:23 [ERR] switch_core_speech.c:60  
> switch_core_speech_open() Invalid speech module [openmrcp]!
> 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859  
> init_speech_engine() Invalid TTS module!
> 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()  Cannot  
> allocate speech engine!
> 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160  
> switch_core_standard_on_execute() Hangup sofia/internal/ 
> 1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
> 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878  
> switch_core_session_thread() Session 2 (sofia/internal/ 
> 1000 at 10.0.0.2) Ended
> 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880  
> switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 
>  [CS_HANGUP]
>
>
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From brian at freeswitch.org  Wed Dec 10 12:27:22 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 14:27:22 -0600
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
Message-ID: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>

FreeSWITCHers,

I'm looking for donations for the next batch of sound files we need to  
have done for the up coming 1.0.2 release.

I have had others pitch in some money in the past and I thank everyone  
for doing so.  I hope everyone
can come together and help me raise about $200 to pay for this batch  
of prompts.

I also would like to thank Bandwidth.com and Teliax for their support  
of the FreeSWITCH project.

Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...

If you wish to donate please paypal brian at freeswitch.org that'll help  
out!

Happy Holidays,
Brian West
FreeSWITCH.org
PS: If you know of any sound files we need let me know.


From mszlazak at aol.com  Wed Dec 10 12:37:25 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 15:37:25 -0500
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
In-Reply-To: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
Message-ID: <8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>


 
How do I donate?


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Cc: freeswitch-dev at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 12:27 pm
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware










FreeSWITCHers,

I'm looking for donations for the next batch of sound files we need to  
have done for the up coming 1.0.2 release.

I have had others pitch in some money in the past and I thank everyone  
for doing so.  I hope everyone
can come together and help me raise about $200 to pay for this batch  
of prompts.

I also would like to thank Bandwidth.com and Teliax for their support  
of the FreeSWITCH project.

Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...

If you wish to donate please paypal brian at freeswitch.org that'll help  
out!

Happy Holidays,
Brian West
FreeSWITCH.org
PS: If you know of any sound files we need let me know.

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From intralanman at freeswitch.org  Wed Dec 10 12:41:27 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 10 Dec 2008 15:41:27 -0500
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>	<87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>
Message-ID: <49402977.4090808@freeswitch.org>

try blocking ICMP packets TO the MSS.... i had this exact same problem a 
few months ago.... MSS starts sending RTP to FS before FS is ready to 
accept.... so the OS catches the port not open and returns an ICMP 3:3 
back to the MSS.... which in turn chokes on the queued up RTP and 
refuses to send anymore...

-Ray

John Rutherford wrote:
> I just emailed it to him.
>
> Thanks!
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
> Sent: Wednesday, December 10, 2008 1:09 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
> On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
> wrote:
>   
>> I have a pcap, but I'm not able to see anything obviously wrong with
>>     
> it.
>
> We find that some equipment (in fact a lot of equipment) have features
> that cause issues to be quite non-obvious, so perhaps you could give
> the pcap to Brian for him to review. He's a total ace when it comes to
> bug hunting.
>
> -MC
>
>   
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>     
> Brian
>   
>> West
>> Sent: Wednesday, December 10, 2008 12:59 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] No audio after transfer
>>
>>
>>
>> would be most helpful to capture a pcap of the entire thing by itself
>>     
> start
>   
>> to finish.
>>
>>
>>
>> /b
>>
>>
>>
>> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>>
>> No. I realize that's it's a B2BUA and that's exactly what we want.
>>
>>
>>
>> Everything with the transfer seems to work fine, except that there is
>>     
> no
>   
>> audio.
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From brian at freeswitch.org  Wed Dec 10 12:43:15 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 14:43:15 -0600
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
In-Reply-To: <8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
Message-ID: <4B1AFF7A-F3DF-41B6-ADF3-14767B44BA41@freeswitch.org>

Paypal works great!  ;)

/b

On Dec 10, 2008, at 2:37 PM, mszlazak at aol.com wrote:

> If you wish to donate please paypal brian at freeswitch.org that'll help

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From intralanman at freeswitch.org  Wed Dec 10 12:45:25 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 10 Dec 2008 15:45:25 -0500
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
In-Reply-To: <8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
Message-ID: <49402A65.7090808@freeswitch.org>

i saw this in the first email

"If you wish to donate please paypal brian at freeswitch.org 
<mailto:brian at freeswitch.org> that'll help "

-Ray


mszlazak at aol.com wrote:
>
> How do I donate?
>
> -----Original Message-----
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Cc: freeswitch-dev at lists.freeswitch.org
> Sent: Wed, 10 Dec 2008 12:27 pm
> Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
>
> FreeSWITCHers,
>
>
>
> I'm looking for donations for the next batch of sound files we need to  
>
> have done for the up coming 1.0.2 release.
>
>
>
> I have had others pitch in some money in the past and I thank everyone  
>
> for doing so.  I hope everyone
>
> can come together and help me raise about $200 to pay for this batch  
>
> of prompts.
>
>
>
> I also would like to thank Bandwidth.com and Teliax for their support  
>
> of the FreeSWITCH project.
>
>
>
> Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...
>
>
>
> If you wish to donate please paypal brian at freeswitch.org <mailto:brian at freeswitch.org> that'll help  
>
> out!
>
>
>
> Happy Holidays,
>
> Brian West
>
> FreeSWITCH.org
>
> PS: If you know of any sound files we need let me know.
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
> Listen to 350+ music, sports, & news radio stations -- including songs 
> for the holidays -- FREE while you browse. Start Listening Now 
> <http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mszlazak at aol.com  Wed Dec 10 13:24:31 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 16:24:31 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com><74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org><8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>
	<D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org>
Message-ID: <8CB2924D2F19E7B-9D4-9EE@Webmail-mg06.sim.aol.com>


 Yup my bad. 

But I'm still getting this error:

2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll
**The specified module could not be found.? **

It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then?

I'm still having the same problem with no audio from Voxeo.

Mark. 


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 11:59 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo









Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.



/b




On Dec 10, 2008, at 1:51 PM, mszlazak at aol.com wrote:



 It was someone from Voxeo support. I think John was the main person helping me with this.
 
 I updated but things got worse all over.
 I now can't run other extensions Gino's pizza or some db stuff
 

 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
?2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!
 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine!
 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000 at 10.0.0.2) Ended
 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]
 
 
 





=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From jaugenstine at gmail.com  Wed Dec 10 13:33:01 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Wed, 10 Dec 2008 13:33:01 -0800
Subject: [Freeswitch-users]  conference module question - prompts
Message-ID: <207e7a5e0812101333k28b27e52tbfe852458eae204f@mail.gmail.com>

I am trying to modify the behavior of the playing of prompts when someone
enters the conference.  When I enable the conf-welcome prompt and a new
participant enters the conference, the prompt is played to the conference
and everyone hears the welcome.  Is there a any way to modify the
configuration so that the welcome prompt is only heard by the participant
entering the conference?

Thank you.
Jonathan
jaugenstine at gmail.com
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From john at loopfx.com  Wed Dec 10 14:18:52 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 17:18:52 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>	<87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>
	<49402977.4090808@freeswitch.org>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>

Okay. I just tried this.  

 

Now we're getting the audio going one way, but not the other.  So, I can
hear the person that I just transferred to, but they can't hear me.

 

Anyone have any other ideas?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Raymond Chandler
Sent: Wednesday, December 10, 2008 3:41 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

try blocking ICMP packets TO the MSS.... i had this exact same problem a
few months ago.... MSS starts sending RTP to FS before FS is ready to
accept.... so the OS catches the port not open and returns an ICMP 3:3
back to the MSS.... which in turn chokes on the queued up RTP and
refuses to send anymore...

-Ray

John Rutherford wrote: 

I just emailed it to him.
 
Thanks!
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Wednesday, December 10, 2008 1:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer
 
On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
<mailto:john at loopfx.com> 
wrote:
  

	I have a pcap, but I'm not able to see anything obviously wrong
with
	    

it.
 
We find that some equipment (in fact a lot of equipment) have features
that cause issues to be quite non-obvious, so perhaps you could give
the pcap to Brian for him to review. He's a total ace when it comes to
bug hunting.
 
-MC
 
  

	 
	 
	From: freeswitch-users-bounces at lists.freeswitch.org
	[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
	    

Brian
  

	West
	Sent: Wednesday, December 10, 2008 12:59 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] No audio after transfer
	 
	 
	 
	would be most helpful to capture a pcap of the entire thing by
itself
	    

start
  

	to finish.
	 
	 
	 
	/b
	 
	 
	 
	On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
	 
	No. I realize that's it's a B2BUA and that's exactly what we
want.
	 
	 
	 
	Everything with the transfer seems to work fine, except that
there is
	    

no
  

	audio.
	 
	 
	 
	_______________________________________________
	Freeswitch-users mailing list
	Freeswitch-users at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	http://www.freeswitch.org
	 
	 
	    

 
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

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From carlos.talbot at gmail.com  Wed Dec 10 14:57:43 2008
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Wed, 10 Dec 2008 16:57:43 -0600
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <8CB2924D2F19E7B-9D4-9EE@Webmail-mg06.sim.aol.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com><74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org><8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>
	<D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org>
	<8CB2924D2F19E7B-9D4-9EE@Webmail-mg06.sim.aol.com>
Message-ID: <587BE2B8-E458-4050-B3AA-46341A2B52B1@gmail.com>

There was a typecast warning that prevented spidermoneky from  
compiling in a recent svn. Did you check to see if it compiled?

Sent from my iPhone

On Dec 10, 2008, at 3:24 PM, mszlazak at aol.com wrote:

> Yup my bad.
>
> But I'm still getting this error:
>
> 2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file()  
> Error Loading module C:\Source\freeswitch-snapshot\Debug\mod 
> \mod_spidermonkey_teletone.dll
> **The specified module could not be found.  **
>
> It doesn't look like it was put in this latest snapshot. I could use  
> that dll from my older snapshot, has it been changed since then?
>
> I'm still having the same problem with no audio from Voxeo.
>
> Mark.
>
>
> -----Original Message-----
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wed, 10 Dec 2008 11:59 am
> Subject: Re: [Freeswitch-users] Audio routing problem between FS and  
> Voxeo
>
> Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.
>
> /b
>
> On Dec 10, 2008, at 1:51 PM, mszlazak at aol.com wrote:
>
>> It was someone from Voxeo support. I think John was the main person  
>> helping me with this.
>>
>> I updated but things got worse all over.
>> I now can't run other extensions Gino's pizza or some db stuff
>>
>>
>> 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57  
>> switch_core_asr_open() Invalid ASR module [pocketsphinx]!
>>  2008-12-10 11:41:23 [ERR] switch_core_speech.c:60  
>> switch_core_speech_open() Invalid speech module [openmrcp]!
>> 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859  
>> init_speech_engine() Invalid TTS module!
>> 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()  Cannot  
>> allocate speech engine!
>> 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160  
>> switch_core_standard_on_execute() Hangup sofia/internal/ 
>> 1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878  
>> switch_core_session_thread() Session 2 (sofia/internal/ 
>> 1000 at 10.0.0.2) Ended
>> 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880  
>> switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 
>>  [CS_HANGUP]
>>
>>
> =
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> Listen to 350+ music, sports, & news radio stations ? including song 
> s for the holidays ? FREE while you browse. Start Listening Now!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Wed Dec 10 14:52:52 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 14:52:52 -0800
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
	<191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
	<BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org>
	<81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>
	<87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>
	<49402977.4090808@freeswitch.org>
	<81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>
Message-ID: <87f2f3b90812101452t7076bb6fo93f7a78bbfb0404f@mail.gmail.com>

I smell a NAT... is there any NAT involved?

On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford <john at loopfx.com> wrote:
> Okay. I just tried this.
>
>
>
> Now we're getting the audio going one way, but not the other.  So, I can
> hear the person that I just transferred to, but they can't hear me.
>
>
>
> Anyone have any other ideas?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raymond
> Chandler
> Sent: Wednesday, December 10, 2008 3:41 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> try blocking ICMP packets TO the MSS.... i had this exact same problem a few
> months ago.... MSS starts sending RTP to FS before FS is ready to accept....
> so the OS catches the port not open and returns an ICMP 3:3 back to the
> MSS.... which in turn chokes on the queued up RTP and refuses to send
> anymore...
>
> -Ray
>
> John Rutherford wrote:
>
> I just emailed it to him.
>
>
>
> Thanks!
>
>
>
> -----Original Message-----
>
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>
> Michael Collins
>
> Sent: Wednesday, December 10, 2008 1:09 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
>
> wrote:
>
>
>
> I have a pcap, but I'm not able to see anything obviously wrong with
>
>
>
> it.
>
>
>
> We find that some equipment (in fact a lot of equipment) have features
>
> that cause issues to be quite non-obvious, so perhaps you could give
>
> the pcap to Brian for him to review. He's a total ace when it comes to
>
> bug hunting.
>
>
>
> -MC
>
>
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>
>
>
> Brian
>
>
>
> West
>
> Sent: Wednesday, December 10, 2008 12:59 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
>
>
>
>
> would be most helpful to capture a pcap of the entire thing by itself
>
>
>
> start
>
>
>
> to finish.
>
>
>
>
>
>
>
> /b
>
>
>
>
>
>
>
> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>
>
>
> No. I realize that's it's a B2BUA and that's exactly what we want.
>
>
>
>
>
>
>
> Everything with the transfer seems to work fine, except that there is
>
>
>
> no
>
>
>
> audio.
>
>
>
>
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mszlazak at aol.com  Wed Dec 10 15:01:11 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 18:01:11 -0500
Subject: [Freeswitch-users]  Audio routing problem between FS and Voxeo
Message-ID: <8CB2932540232CF-9D4-1024@Webmail-mg06.sim.aol.com>

Brian, I'm still having the same audio problem when bridging/routing to Voxeo using the latest snapshot. Help!





Also, it doesn't look like mod_spidermonkey_teletone.dll was
put into this latest snapshot. Has it been changed since November? I could use that older version of this dll.











Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 





 


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From brian at freeswitch.org  Wed Dec 10 15:06:17 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 17:06:17 -0600
Subject: [Freeswitch-users] Hardware requests.
Message-ID: <1B94AB6C-022D-47E8-9868-92CE89EA0FD4@freeswitch.org>

Does anyone have a Polycom 320 they would like to donate to the  
project?  I need one to trouble shoot a problem.  Please contact me  
off list.

Thanks,
Brian West



From mszlazak at aol.com  Wed Dec 10 15:48:01 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 18:48:01 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <587BE2B8-E458-4050-B3AA-46341A2B52B1@gmail.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com><74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org><8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com><D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org><8CB2924D2F19E7B-9D4-9EE@Webmail-mg06.sim.aol.com>
	<587BE2B8-E458-4050-B3AA-46341A2B52B1@gmail.com>
Message-ID: <8CB2938DF2CC0FD-9D4-12BD@Webmail-mg06.sim.aol.com>

I only glanced at the
compilers output when it ended and it reported no no errors but I did
not look at the warnings along the way. Have things been modified since
so spider monkey compiles?? 



More importantly, do you have any ideas as what is going on with my audio problem. I can't attach a wireshark .pcap file since it's to big for your list and my email gets rejected.

 


 

-----Original Message-----
From: Carlos Talbot <carlos.talbot at gmail.com>
To: freeswitch-users at lists.freeswitch.org <freeswitch-users at lists.freeswitch.org>
Cc: freeswitch-users at lists.freeswitch.org <freeswitch-users at lists.freeswitch.org>
Sent: Wed, 10 Dec 2008 2:57 pm
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo










There was a typecast warning that prevented spidermoneky from compiling in a recent svn. Did you check to see if it compiled?

Sent from my iPhone


On Dec 10, 2008, at 3:24 PM, mszlazak at aol.com wrote:








 Yup my bad. 



But I'm still getting this error:



2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll

**The specified module could not be found.? **



It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then?



I'm still having the same problem with no audio from Voxeo.



Mark. 





 





 



-----Original Message-----

From: Brian West <brian at frees
witch.org>

To: freeswitch-users at lists.freeswitch.org

Sent: Wed, 10 Dec 2008 11:59 am

Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo












Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.







/b









On Dec 10, 2008, at 1:51 PM, mszlazak at aol.com wrote:





 It was someone from Voxeo support. I think John was the main person helping me with this.

 

 I updated but things got worse all over.

 I now can't run other extensions Gino's pizza or some db stuff

 



 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!

?2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!

 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!

 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine!

 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]

 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000 at 10.0.0.2) Ended

 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]

 

 

 










=



 






_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitc
h-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





 





Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From anthony.minessale at gmail.com  Wed Dec 10 16:48:19 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Dec 2008 18:48:19 -0600
Subject: [Freeswitch-users] conference module question - prompts
In-Reply-To: <207e7a5e0812101333k28b27e52tbfe852458eae204f@mail.gmail.com>
References: <207e7a5e0812101333k28b27e52tbfe852458eae204f@mail.gmail.com>
Message-ID: <191c3a030812101648k53435801iaa3e4a3ba26560e5@mail.gmail.com>

not currently,  no

On Wed, Dec 10, 2008 at 3:33 PM, jonathan augenstine
<jaugenstine at gmail.com>wrote:

> I am trying to modify the behavior of the playing of prompts when someone
> enters the conference.  When I enable the conf-welcome prompt and a new
> participant enters the conference, the prompt is played to the conference
> and everyone hears the welcome.  Is there a any way to modify the
> configuration so that the welcome prompt is only heard by the participant
> entering the conference?
>
> Thank you.
> Jonathan
> jaugenstine at gmail.com
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From krice at suspicious.org  Wed Dec 10 17:39:32 2008
From: krice at suspicious.org (Ken Rice)
Date: Wed, 10 Dec 2008 19:39:32 -0600
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for pending
	1.0.2/Hardware
In-Reply-To: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
Message-ID: <C565CB74.4AFF2%krice@suspicious.org>

Hey Guys... I spoke with Brian on this a few minutes ago and some money has
already showed up for the sound files... Lets see if we can go ahead and
make it where Brian can get these files on order early tomorrow so get can
make sure they get us a good Christmas Present in the form of a new stable
release

K 


> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Wed, 10 Dec 2008 14:27:22 -0600
> To: <freeswitch-users at lists.freeswitch.org>
> Cc: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: [Freeswitch-dev] Sounds for pending 1.0.2/Hardware
> 
> FreeSWITCHers,
> 
> I'm looking for donations for the next batch of sound files we need to
> have done for the up coming 1.0.2 release.
> 
> I have had others pitch in some money in the past and I thank everyone
> for doing so.  I hope everyone
> can come together and help me raise about $200 to pay for this batch
> of prompts.
> 
> I also would like to thank Bandwidth.com and Teliax for their support
> of the FreeSWITCH project.
> 
> Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...
> 
> If you wish to donate please paypal brian at freeswitch.org that'll help
> out!
> 
> Happy Holidays,
> Brian West
> FreeSWITCH.org
> PS: If you know of any sound files we need let me know.
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org




From ack at telefonica.net  Wed Dec 10 17:51:56 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Thu, 11 Dec 2008 02:51:56 +0100
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for
	pending	1.0.2/Hardware
In-Reply-To: <C565CB74.4AFF2%krice@suspicious.org>
References: <C565CB74.4AFF2%krice@suspicious.org>
Message-ID: <1228960316.10071.21.camel@develop4>

I'm in too . Brian hope you got money i sent, a pleasure to contribute.

Cheers, ( sack )

El mi?, 10-12-2008 a las 19:39 -0600, Ken Rice escribi?:
> Hey Guys... I spoke with Brian on this a few minutes ago and some money has
> already showed up for the sound files... Lets see if we can go ahead and
> make it where Brian can get these files on order early tomorrow so get can
> make sure they get us a good Christmas Present in the form of a new stable
> release
> 
> K 
> 
> 
> > From: Brian West <brian at freeswitch.org>
> > Reply-To: "freeswitch-dev at lists.freeswitch.org"
> > <freeswitch-dev at lists.freeswitch.org>
> > Date: Wed, 10 Dec 2008 14:27:22 -0600
> > To: <freeswitch-users at lists.freeswitch.org>
> > Cc: "freeswitch-dev at lists.freeswitch.org"
> > <freeswitch-dev at lists.freeswitch.org>
> > Subject: [Freeswitch-dev] Sounds for pending 1.0.2/Hardware
> > 
> > FreeSWITCHers,
> > 
> > I'm looking for donations for the next batch of sound files we need to
> > have done for the up coming 1.0.2 release.
> > 
> > I have had others pitch in some money in the past and I thank everyone
> > for doing so.  I hope everyone
> > can come together and help me raise about $200 to pay for this batch
> > of prompts.
> > 
> > I also would like to thank Bandwidth.com and Teliax for their support
> > of the FreeSWITCH project.
> > 
> > Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...
> > 
> > If you wish to donate please paypal brian at freeswitch.org that'll help
> > out!
> > 
> > Happy Holidays,
> > Brian West
> > FreeSWITCH.org
> > PS: If you know of any sound files we need let me know.
> > 
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

"No basta saber, hay que aplicar lo que se sabe; 
no basta querer hacerlas cosas, hay que hacerlas".

"Knowing is not enough; we must apply. 
 Willing is not enough; we must do"

 Johann Wolfgang von Goethe
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From brian at freeswitch.org  Wed Dec 10 17:56:33 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 19:56:33 -0600
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for
	pending	1.0.2/Hardware
In-Reply-To: <1228960316.10071.21.camel@develop4>
References: <C565CB74.4AFF2%krice@suspicious.org>
	<1228960316.10071.21.camel@develop4>
Message-ID: <C114E867-AE3B-411F-912B-A3B203C4AF8A@freeswitch.org>

Thank you it really helps.  I want to make sure the 1.0.2 release is  
the best release ever!

/b

On Dec 10, 2008, at 7:51 PM, Angel Carpintero wrote:

> I'm in too . Brian hope you got money i sent, a pleasure to  
> contribute.



From mszlazak at aol.com  Wed Dec 10 18:23:57 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 21:23:57 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for pending
	1.0.2/Hardware
In-Reply-To: <C114E867-AE3B-411F-912B-A3B203C4AF8A@freeswitch.org>
References: <C565CB74.4AFF2%krice@suspicious.org><1228960316.10071.21.camel@develop4>
	<C114E867-AE3B-411F-912B-A3B203C4AF8A@freeswitch.org>
Message-ID: <8CB294EA800ADED-9D4-1AE7@Webmail-mg06.sim.aol.com>


 Good enough will do to get my cash. 
It will be on the way once my paypal account is confirmed in a few days.


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 5:56 pm
Subject: Re: [Freeswitch-users] [Freeswitch-dev] Sounds for pending 1.0.2/Hardware










Thank you it really helps.  I want to make sure the 1.0.2 release is  
the best release ever!

/b

On Dec 10, 2008, at 7:51 PM, Angel Carpintero wrote:

> I'm in too . Brian hope you got money i sent, a pleasure to  
> contribute.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From edpimentl at gmail.com  Wed Dec 10 19:49:55 2008
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 10 Dec 2008 22:49:55 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for pending
	1.0.2/Hardware
In-Reply-To: <C565CB74.4AFF2%krice@suspicious.org>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<C565CB74.4AFF2%krice@suspicious.org>
Message-ID: <9dc4a1670812101949m7864aeb9m1bb8b069cff8fadf@mail.gmail.com>

Bryan!!,

Count on me too.
E
http://Gpro.ws
http://DatR.ws (Store, Sync, Share, Publish)
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From edpimentl at gmail.com  Wed Dec 10 20:41:22 2008
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 10 Dec 2008 23:41:22 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for pending
	1.0.2/Hardware
In-Reply-To: <C565CB74.4AFF2%krice@suspicious.org>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<C565CB74.4AFF2%krice@suspicious.org>
Message-ID: <9dc4a1670812102041p4c5e2af8i77ae5d8ef4a9f1c@mail.gmail.com>

Here is some "sound advice" regarding some unique sounds links:

http://flashkit.com/soundfx/
Flash Kit has an ever growing list of shareware and freeware SoundFX for
download. It provides a powerful searching, excellent organisation and easy
real time  previews make this the most advanced Sound FX download site on
the net!  mp3 and flashtrak versions of all effects as well!

http://www.audiomicro.com/royalty-free-sound-effects.html
AudioMicro is a revolutionary collection of user-generated micro stock
music, sound effects, production music, production elements and music  cues.
Finally, high quality audio content is available at unbelievable  prices
with no hidden costs or fees.

http://www.sound-effect.com/
Search for, preview and download royalty free sound effects for immediate
use in your multimedia project. These high quality royalty free sound
effects are hand-picked from only the best sound designers

http://www.soundsnap.com/browse
Soundsnap is the best platform to find and share free sound effects and
loops- legally. It is a collection of original sounds made or recorded  by
its users, and not songs or sound FX found on commercial libraries  or
sample CD


http://www.findsounds.com/
FindSounds.com is a free site for finding sound effects and musical
instrument samples on the Web. It is a Web search engine, like Google and
AltaVista, but with a focus on sounds. It provides powerful features, yet is
simple and easy to use, and suitable for all ages

http://soungle.com/
Soungle is a free site, developed by Southern Codes, for finding all  kind
of sound FX and musical instruments samples on our mega online  library. As
different from most of similar sites, Soungle is NOT a Web  search engine.
It only searches in our growing monster database. Our  goals are to keep it
simple to use (search, preview and download) and  to keep it free

http://www.sfxsource.com/Sound-Effects
In SFX Source youll find cutting-edge and imaginative sound samples crafted
with  passionate expertise for use in all levels of production, from
professional to amateur, for use in Film, TV, Games, and New Media.


http://www.a1freesoundeffects.com/
A1 Free Sound Effects wants to provide the internet with our Free Sound
Effects that you can download to your computer and use for church, school,
home or for any non-profit project. Commercial Sounds Available.


http://soundrangers.com/html/free-sound-effects.html
Soundrangers specializes in generating state-of-the-art royalty free  sound
effects and music for interactive media, such as virtual  user-interfaces,
games, online entertainment, web sites and  communication devices.

http://www.partnersinrhyme.com/pir/PIRsfx.shtml
Partners In Rhyme provide free  resources, help and advice to amateur and
professional multimedia  producers, film makers, musicians and students in
their search for  music, sound effects and audio tools to complete their
projects  inexpensively and quickly

http://www.soundboard.com/
Soundboard.com puts all your audio clips, as well as millions from other
users in one, easy to manage location.  Its mission is to invite everyone to
help us create a central site for audio clips in a format that anyone with a
connection and a browser can enjoy

E
http://Gpro.ws
http://DatR.ws (Store, Sync, Share, Publish)
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From carole.olivier at enst.fr  Wed Dec 10 22:51:41 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 10 Dec 2008 22:51:41 -0800 (PST)
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
Message-ID: <20950299.post@talk.nabble.com>


Hello,

I am running the version 1.0.1.
Do you still need me to run the debugging?

Carole


Brian West-3 wrote:
> 
> I have already labbed this up on SVN trunk and I don't get a segfault  
> but I get something else that prevents it from working properly.    We  
> are working on it today.  Also what version are you running?
> 
> /b
> 
> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
> 
>> Thanks for reporting this. It would be helpful to know a bit more. Can
>> you start freeswitch and press F12 (or type "version" at the CLI) and
>> report back what it says?
>> Also, a backtrace (bt) is generally useful. If you could produce a
>> "bt" and a "bt full" from you core file that would be extremely
>> helpful.
>>
>> see this link for more information:
>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>
>> you should have a "core" file for each segfault that occurred. Use the
>> gdb program to get the back trace:
>>
>> gdb /path/to/fs/binary core.xxx
>>
>> then capture the output from these two commands:
>>
>> bt <enter>
>> bt full <enter>
>>
>> When you type those commands you'll see tons of debugging info;
>> capture that and put it in a pastebin (pastebin.freeswitch.org) then
>> report back here.
>>
>> You can exit the gdb debugger by typing q <enter>
>>
>> Thanks for helping us collect information!
>>
>> -MC
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Wed Dec 10 23:05:16 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 23:05:16 -0800
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20950299.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
Message-ID: <87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>

Carole,

There have been many updates since 1.0.1 was officially released. If
you could start FreeSWITCH and then press F12 it will reveal which SVN
revisionnumber you are running. Please supply that number and it will
help us to know if you are on a recent revision.

Thanks,
MC

On Wed, Dec 10, 2008 at 10:51 PM, Carole O. <carole.olivier at enst.fr> wrote:
>
> Hello,
>
> I am running the version 1.0.1.
> Do you still need me to run the debugging?
>
> Carole
>
>
> Brian West-3 wrote:
>>
>> I have already labbed this up on SVN trunk and I don't get a segfault
>> but I get something else that prevents it from working properly.    We
>> are working on it today.  Also what version are you running?
>>
>> /b
>>
>> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
>>
>>> Thanks for reporting this. It would be helpful to know a bit more. Can
>>> you start freeswitch and press F12 (or type "version" at the CLI) and
>>> report back what it says?
>>> Also, a backtrace (bt) is generally useful. If you could produce a
>>> "bt" and a "bt full" from you core file that would be extremely
>>> helpful.
>>>
>>> see this link for more information:
>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>
>>> you should have a "core" file for each segfault that occurred. Use the
>>> gdb program to get the back trace:
>>>
>>> gdb /path/to/fs/binary core.xxx
>>>
>>> then capture the output from these two commands:
>>>
>>> bt <enter>
>>> bt full <enter>
>>>
>>> When you type those commands you'll see tons of debugging info;
>>> capture that and put it in a pastebin (pastebin.freeswitch.org) then
>>> report back here.
>>>
>>> You can exit the gdb debugger by typing q <enter>
>>>
>>> Thanks for helping us collect information!
>>>
>>> -MC
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> --
> View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From carole.olivier at enst.fr  Wed Dec 10 23:33:29 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 10 Dec 2008 23:33:29 -0800 (PST)
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
	<87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>
Message-ID: <20950711.post@talk.nabble.com>


I have got the following:
Freeswitch Version 1.0.1 (9171)

Carole



Michael Collins-11 wrote:
> 
> Carole,
> 
> There have been many updates since 1.0.1 was officially released. If
> you could start FreeSWITCH and then press F12 it will reveal which SVN
> revisionnumber you are running. Please supply that number and it will
> help us to know if you are on a recent revision.
> 
> Thanks,
> MC
> 
> On Wed, Dec 10, 2008 at 10:51 PM, Carole O. <carole.olivier at enst.fr>
> wrote:
>>
>> Hello,
>>
>> I am running the version 1.0.1.
>> Do you still need me to run the debugging?
>>
>> Carole
>>
>>
>> Brian West-3 wrote:
>>>
>>> I have already labbed this up on SVN trunk and I don't get a segfault
>>> but I get something else that prevents it from working properly.    We
>>> are working on it today.  Also what version are you running?
>>>
>>> /b
>>>
>>> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
>>>
>>>> Thanks for reporting this. It would be helpful to know a bit more. Can
>>>> you start freeswitch and press F12 (or type "version" at the CLI) and
>>>> report back what it says?
>>>> Also, a backtrace (bt) is generally useful. If you could produce a
>>>> "bt" and a "bt full" from you core file that would be extremely
>>>> helpful.
>>>>
>>>> see this link for more information:
>>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>>
>>>> you should have a "core" file for each segfault that occurred. Use the
>>>> gdb program to get the back trace:
>>>>
>>>> gdb /path/to/fs/binary core.xxx
>>>>
>>>> then capture the output from these two commands:
>>>>
>>>> bt <enter>
>>>> bt full <enter>
>>>>
>>>> When you type those commands you'll see tons of debugging info;
>>>> capture that and put it in a pastebin (pastebin.freeswitch.org) then
>>>> report back here.
>>>>
>>>> You can exit the gdb debugger by typing q <enter>
>>>>
>>>> Thanks for helping us collect information!
>>>>
>>>> -MC
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> --
>> View this message in context:
>> http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950711.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Wed Dec 10 23:58:44 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 10 Dec 2008 23:58:44 -0800
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20950711.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
	<87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>
	<20950711.post@talk.nabble.com>
Message-ID: <7DF34003-ACE0-496A-B986-E91E83C75C4B@freeswitch.org>

Wow! That is really old. I strongly recommend that you update to the  
latest trunk using svn.

-MC


On Dec 10, 2008, at 11:33 PM, "Carole O." <carole.olivier at enst.fr>  
wrote:

>
> I have got the following:
> Freeswitch Version 1.0.1 (9171)
>
> Carole
>
>
>
> Michael Collins-11 wrote:
>>
>> Carole,
>>
>> There have been many updates since 1.0.1 was officially released. If
>> you could start FreeSWITCH and then press F12 it will reveal which  
>> SVN
>> revisionnumber you are running. Please supply that number and it will
>> help us to know if you are on a recent revision.
>>
>> Thanks,
>> MC
>>
>> On Wed, Dec 10, 2008 at 10:51 PM, Carole O. <carole.olivier at enst.fr>
>> wrote:
>>>
>>> Hello,
>>>
>>> I am running the version 1.0.1.
>>> Do you still need me to run the debugging?
>>>
>>> Carole
>>>
>>>
>>> Brian West-3 wrote:
>>>>
>>>> I have already labbed this up on SVN trunk and I don't get a  
>>>> segfault
>>>> but I get something else that prevents it from working  
>>>> properly.    We
>>>> are working on it today.  Also what version are you running?
>>>>
>>>> /b
>>>>
>>>> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
>>>>
>>>>> Thanks for reporting this. It would be helpful to know a bit  
>>>>> more. Can
>>>>> you start freeswitch and press F12 (or type "version" at the  
>>>>> CLI) and
>>>>> report back what it says?
>>>>> Also, a backtrace (bt) is generally useful. If you could produce a
>>>>> "bt" and a "bt full" from you core file that would be extremely
>>>>> helpful.
>>>>>
>>>>> see this link for more information:
>>>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>>>
>>>>> you should have a "core" file for each segfault that occurred.  
>>>>> Use the
>>>>> gdb program to get the back trace:
>>>>>
>>>>> gdb /path/to/fs/binary core.xxx
>>>>>
>>>>> then capture the output from these two commands:
>>>>>
>>>>> bt <enter>
>>>>> bt full <enter>
>>>>>
>>>>> When you type those commands you'll see tons of debugging info;
>>>>> capture that and put it in a pastebin (pastebin.freeswitch.org)  
>>>>> then
>>>>> report back here.
>>>>>
>>>>> You can exit the gdb debugger by typing q <enter>
>>>>>
>>>>> Thanks for helping us collect information!
>>>>>
>>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950711.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From yudha2008 at gmail.com  Thu Dec 11 00:51:09 2008
From: yudha2008 at gmail.com (Baskar)
Date: Thu, 11 Dec 2008 14:21:09 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <191c3a030812030924u457f934ep77bd70680f583fcd@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
	<d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
	<191c3a030812030924u457f934ep77bd70680f583fcd@mail.gmail.com>
Message-ID: <d0640d380812110051w41bc5481r60a3d3242044d3f7@mail.gmail.com>

*Hi,

    when in dial from console inbound is working fine when i dial outbound
it is not working in console dialing.
*

FreeSWITCH Version 1.0.trunk (10567) Started.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
2008-12-11 14:17:03 [CONSOLE] mod_local_stream.c:142 read_stream_thread()
Can't open directory: /usr/local/freeswitch/sounds/music/16000

*freeswitch at localhost> pa devlist*
API CALL [pa(devlist)] output:
0;/dev/dsp;16;4
1;Intel ICH5: Intel ICH5 (hw:0,0);2;6
2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0
3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0
4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0
5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2
6;front;0;6
7;surround40;0;4
8;surround41;0;128
9;surround50;0;128
10;surround51;0;6
11;iec958;0;2
12;spdif;0;2
13;default;128;128
14;dmix;0;2
*
After that i dial 3 in softphone
Output:*

freeswitch at localhost> 2008-12-11 14:17:16 [NOTICE] switch_channel.c:564
switch_channel_set_name() New Channel
sofia/internal/1003 at 172.20.177.117[d49468e5-be90-40f6-8ffe-58c56651d87a]
2008-12-11 14:17:16 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
FreeSwitch->3 in context default
2008-12-11 14:17:16 [NOTICE] switch_channel.c:564 switch_channel_set_name()
New Channel portaudio/1003 [bfc0023c-0725-40a8-a187-574bdab40c40]
2008-12-11 14:17:17 [NOTICE] mod_portaudio.c:235 channel_on_init()
Ring-Ready portaudio/1003!
*pa answer*
2008-12-11 14:17:24 [NOTICE] mod_portaudio.c:1404 answer_call() Channel
[portaudio/1003] has been answered
API CALL [pa(answer)] output:
Answered 1 channels.

freeswitch at localhost> 2008-12-11 14:17:24 [NOTICE]
switch_ivr_originate.c:1509 switch_ivr_originate() Channel [sofia/internal/
1003 at 172.20.177.117] has been answered
pa hangup
2008-12-11 14:17:32 [NOTICE] mod_portaudio.c:1365 hangup_call() Hangup
portaudio/1003 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
API CALL [pa(hangup)] output:
OK

freeswitch at localhost> 2008-12-11 14:17:32 [NOTICE]
switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup
sofia/internal/1003 at 172.20.177.117 [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-11 14:17:32 [INFO] mod_cdr_csv.c:207 my_on_hangup() CHANNEL_DATA:
Channel-State: [CS_HANGUP]
Channel-State-Number: [10]
Channel-Name: [sofia/internal/1003 at 172.20.177.117]
Unique-ID: [d49468e5-be90-40f6-8ffe-58c56651d87a]
Call-Direction: [inbound]
Answer-State: [answered]
Caller-Username: [1003]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [FreeSwitch]
Caller-Caller-ID-Number: [1003]
Caller-Network-Addr: [172.20.177.201]
Caller-Destination-Number: [3]
Caller-Unique-ID: [d49468e5-be90-40f6-8ffe-58c56651d87a]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/1003 at 172.20.177.117]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1228985236919311]
Caller-Channel-Created-Time: [1228985236919311]
Caller-Channel-Answered-Time: [1228985244720686]
Caller-Channel-Progress-Time: [1228985237539254]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [1228985252428470]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
Other-Leg-Username: [1003]
Other-Leg-Dialplan: [XML]
Other-Leg-Caller-ID-Name: [Extension 1003]
Other-Leg-Caller-ID-Number: [1003]
Other-Leg-Network-Addr: [172.20.177.201]
Other-Leg-Unique-ID: [bfc0023c-0725-40a8-a187-574bdab40c40]
Other-Leg-Source: [mod_sofia]
Other-Leg-Context: [default]
Other-Leg-Channel-Name: [portaudio/1003]
Other-Leg-Screen-Bit: [true]
Other-Leg-Privacy-Hide-Name: [false]
Other-Leg-Privacy-Hide-Number: [false]


2008-12-11 14:17:32 [NOTICE] switch_core_session.c:956
switch_core_session_thread() Session 1 (sofia/internal/1003 at 172.20.177.117)
Ended
2008-12-11 14:17:32 [NOTICE] switch_core_session.c:958
switch_core_session_thread() Close Channel sofia/internal/
1003 at 172.20.177.117 [CS_HANGUP]
2008-12-11 14:17:32 [NOTICE] switch_core_session.c:956
switch_core_session_thread() Session 2 (portaudio/1003) Ended
2008-12-11 14:17:32 [NOTICE] switch_core_session.c:958
switch_core_session_thread() Close Channel portaudio/1003 [CS_HANGUP]
*
Then i tried it for outbound
Output:*

*pa call 1003*
2008-12-11 14:17:39 [NOTICE] switch_channel.c:564 switch_channel_set_name()
New Channel portaudio/1003 [14de48f4-40cb-42bb-8f43-084d4df7ec89]
2008-12-11 14:17:39 [NOTICE] mod_portaudio.c:1586 place_call() Channel
[portaudio/1003] has been answered
API CALL [pa(call 1003)] output:
SUCCESS:2:14de48f4-40cb-42bb-8f43-084d4df7ec89

freeswitch at localhost> 2008-12-11 14:17:39 [INFO] mod_dialplan_xml.c:233
dialplan_hunt() Processing FreeSWITCH->1003 in context default
2008-12-11 14:17:39 [ERR] mod_sofia.c:2102 sofia_outgoing_channel() Invalid
Gateway
2008-12-11 14:17:39 [NOTICE] mod_sofia.c:2301 sofia_outgoing_channel() Close
Channel N/A [CS_NEW]
2008-12-11 14:17:39 [ERR] switch_ivr_originate.c:1063 switch_ivr_originate()
Cannot create outgoing channel of type [sofia] cause:
[INVALID_NUMBER_FORMAT]
2008-12-11 14:17:39 [INFO] mod_dptools.c:1868 audio_bridge_function()
Originate Failed.  Cause: INVALID_NUMBER_FORMAT
2008-12-11 14:17:39 [NOTICE] mod_dptools.c:1895 audio_bridge_function()
Hangup portaudio/1003 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2008-12-11 14:17:39 [INFO] mod_cdr_csv.c:207 my_on_hangup() CHANNEL_DATA:
Channel-State: [CS_HANGUP]
Channel-State-Number: [10]
Channel-Name: [portaudio/1003]
Unique-ID: [14de48f4-40cb-42bb-8f43-084d4df7ec89]
Call-Direction: [inbound]
Answer-State: [answered]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [FreeSWITCH]
Caller-Caller-ID-Number: [0000000000]
Caller-Network-Addr: [172.20.177.117]
Caller-Destination-Number: [1003]
Caller-Unique-ID: [14de48f4-40cb-42bb-8f43-084d4df7ec89]
Caller-Source: [mod_portaudio]
Caller-Context: [default]
Caller-Channel-Name: [portaudio/1003]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1228985259280611]
Caller-Channel-Created-Time: [1228985259280611]
Caller-Channel-Answered-Time: [1228985259508649]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [1228985259512660]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]


2008-12-11 14:17:39 [NOTICE] switch_core_session.c:956
switch_core_session_thread() Session 3 (portaudio/1003) Ended
2008-12-11 14:17:39 [NOTICE] switch_core_session.c:958
switch_core_session_thread() Close Channel portaudio/1003 [CS_HANGUP]

*Correct me were in am wrong . i have done all the updates and i install
freeswitch newly .

I am using Centos 5.2

I also attached the default.xml in this mail.

Correct me were in am wrong.

-- 
Thanks,
N.Baskar*
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From carole.olivier at enst.fr  Thu Dec 11 03:14:12 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 11 Dec 2008 03:14:12 -0800 (PST)
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <7DF34003-ACE0-496A-B986-E91E83C75C4B@freeswitch.org>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
	<87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>
	<20950711.post@talk.nabble.com>
	<7DF34003-ACE0-496A-B986-E91E83C75C4B@freeswitch.org>
Message-ID: <20953570.post@talk.nabble.com>


ok !

Well, I installed it from opensuse.org, I thought it would be easier for me
since I am completely new here.
Is there a simple way to update this package or would you recommend me to
uninstall the rpm and install freeswitch completely new from the source code
you provide?

Thanks,
Carole


Michael Collins-11 wrote:
> 
> Wow! That is really old. I strongly recommend that you update to the  
> latest trunk using svn.
> 
> -MC
> 
> 
> On Dec 10, 2008, at 11:33 PM, "Carole O." <carole.olivier at enst.fr>  
> wrote:
> 
>>
>> I have got the following:
>> Freeswitch Version 1.0.1 (9171)
>>
>> Carole
>>
>>
>>
>> Michael Collins-11 wrote:
>>>
>>> Carole,
>>>
>>> There have been many updates since 1.0.1 was officially released. If
>>> you could start FreeSWITCH and then press F12 it will reveal which  
>>> SVN
>>> revisionnumber you are running. Please supply that number and it will
>>> help us to know if you are on a recent revision.
>>>
>>> Thanks,
>>> MC
>>>
>>> On Wed, Dec 10, 2008 at 10:51 PM, Carole O. <carole.olivier at enst.fr>
>>> wrote:
>>>>
>>>> Hello,
>>>>
>>>> I am running the version 1.0.1.
>>>> Do you still need me to run the debugging?
>>>>
>>>> Carole
>>>>
>>>>
>>>> Brian West-3 wrote:
>>>>>
>>>>> I have already labbed this up on SVN trunk and I don't get a  
>>>>> segfault
>>>>> but I get something else that prevents it from working  
>>>>> properly.    We
>>>>> are working on it today.  Also what version are you running?
>>>>>
>>>>> /b
>>>>>
>>>>> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
>>>>>
>>>>>> Thanks for reporting this. It would be helpful to know a bit  
>>>>>> more. Can
>>>>>> you start freeswitch and press F12 (or type "version" at the  
>>>>>> CLI) and
>>>>>> report back what it says?
>>>>>> Also, a backtrace (bt) is generally useful. If you could produce a
>>>>>> "bt" and a "bt full" from you core file that would be extremely
>>>>>> helpful.
>>>>>>
>>>>>> see this link for more information:
>>>>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>>>>
>>>>>> you should have a "core" file for each segfault that occurred.  
>>>>>> Use the
>>>>>> gdb program to get the back trace:
>>>>>>
>>>>>> gdb /path/to/fs/binary core.xxx
>>>>>>
>>>>>> then capture the output from these two commands:
>>>>>>
>>>>>> bt <enter>
>>>>>> bt full <enter>
>>>>>>
>>>>>> When you type those commands you'll see tons of debugging info;
>>>>>> capture that and put it in a pastebin (pastebin.freeswitch.org)  
>>>>>> then
>>>>>> report back here.
>>>>>>
>>>>>> You can exit the gdb debugger by typing q <enter>
>>>>>>
>>>>>> Thanks for helping us collect information!
>>>>>>
>>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> --
>>>> View this message in context:
>>>> http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> -- 
>> View this message in context:
>> http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950711.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20953570.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From william at channelxstream.com  Wed Dec 10 23:01:02 2008
From: william at channelxstream.com (William King)
Date: Wed, 10 Dec 2008 23:01:02 -0800
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20950299.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
Message-ID: <1228978862.31447.5.camel@quentusrex-desktop>

Carole,

1.0.1 is known to be broken now. Can you go into the directory where you
have the freeswitch source from the svn repo and type 'make current'.
This will update freeswitch to trunk.

-William 

On Wed, 2008-12-10 at 22:51 -0800, Carole O. wrote:
> Hello,
> 
> I am running the version 1.0.1.
> Do you still need me to run the debugging?
> 
> Carole
> 
> 
> Brian West-3 wrote:
> > 
> > I have already labbed this up on SVN trunk and I don't get a segfault  
> > but I get something else that prevents it from working properly.    We  
> > are working on it today.  Also what version are you running?
> > 
> > /b
> > 
> > On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
> > 
> >> Thanks for reporting this. It would be helpful to know a bit more. Can
> >> you start freeswitch and press F12 (or type "version" at the CLI) and
> >> report back what it says?
> >> Also, a backtrace (bt) is generally useful. If you could produce a
> >> "bt" and a "bt full" from you core file that would be extremely
> >> helpful.
> >>
> >> see this link for more information:
> >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
> >>
> >> you should have a "core" file for each segfault that occurred. Use the
> >> gdb program to get the back trace:
> >>
> >> gdb /path/to/fs/binary core.xxx
> >>
> >> then capture the output from these two commands:
> >>
> >> bt <enter>
> >> bt full <enter>
> >>
> >> When you type those commands you'll see tons of debugging info;
> >> capture that and put it in a pastebin (pastebin.freeswitch.org) then
> >> report back here.
> >>
> >> You can exit the gdb debugger by typing q <enter>
> >>
> >> Thanks for helping us collect information!
> >>
> >> -MC
> > 
> > 
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> 
-- 
William King
Cell: 253-686-5518
E-mail: william at channelxstream.com
Channel XStream
www.channelxstream.com
1-877-600-6786

If there is a possibility that any information in our conversation might
be considered 'private' or 'sensitive' such as passwords, account 
information, legal or financial information, or anything else that you
would consider 'private' or 'sensitive' communications. It is better to 
always err on the side of security. Please encrypt the e-mail using 
my gpg key: 95C9D5B3.

If you are unfamiliar with e-mail encryption feel free to let me know 
and I can help you establish the proper protocols and procedures.
https://help.ubuntu.com/community/GnuPrivacyGuardHowto

Get my gpg key: gpg --recv-key --keyserver keyserver.ubuntu.com 95C9D5B3
Key Fingerprint: EA6F B2EE 1846 55D4 FFD9  80BA 6489 B48C 95C9 D5B3
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From alex at sinapticode.ro  Thu Dec 11 01:21:07 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 11:21:07 +0200
Subject: [Freeswitch-users] Configuring FreeSwitch
Message-ID: <1228987267.4843.6.camel@gathern.lan>

Hi,

I'm a newbie trying to configure freeswitch.

Our needs are simple: for starters we need to call external phones from
freeswitch and play a wav file. 

Now, we have the voice provider's IP (no user/password authentication),
and we have the codec used (called "ulaw" in Asterisk, which I think is
called PCMU?). And this is basically all that we specify in Asterisk.

I compiled Freeswitch with the default settings, and I was wondering if
you guys can give me some hits, point me in the right direction.

Thanks,

-- 
Alexandru Nedelcu
http://alexn.org



From mike at jerris.com  Thu Dec 11 03:32:37 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Dec 2008 06:32:37 -0500
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812110051w41bc5481r60a3d3242044d3f7@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
	<d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
	<191c3a030812030924u457f934ep77bd70680f583fcd@mail.gmail.com>
	<d0640d380812110051w41bc5481r60a3d3242044d3f7@mail.gmail.com>
Message-ID: <4A5A4648-37F4-4887-AD71-005998A7E8CD@jerris.com>

Check your dialplan, you have a gateway name that is not configured  
properly.


On Dec 11, 2008, at 3:51 AM, Baskar wrote:

> 2008-12-11 14:17:39 [ERR] mod_sofia.c:2102 sofia_outgoing_channel()  
> Invalid Gateway



From alex at sinapticode.ro  Thu Dec 11 03:55:39 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 13:55:39 +0200
Subject: [Freeswitch-users] Freeswitch Dialer configuration
Message-ID: <1228996539.4843.13.camel@gathern.lan>

Hi,

I'm trying to setup a simple dialer with Freeswitch.

What I have right now is the following dialplan:

  <extension name="wavs">
    <condition field="destination_number" expression="^2009$">
      <action application="playback"
data="/var/lib/axonite/factura.wav"/>
    </condition>
  </extension>

Right now I'm testing the setup with the following command:
  originate sofia/external/123123123123 at provider.com 2009

How can I configure it (or where to find examples) for the following:
1) the message should start when the phone is answered (right now it
starts when the phone starts ringing I think)
2) I need keys interaction ... like when the user presses 1, the message
should repeat itself, and when the user presses 2 another message should
play

Thank you,

-- 
Alexandru Nedelcu
Software Developer, Sinapticode



From carole.olivier at enst.fr  Thu Dec 11 05:13:47 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 11 Dec 2008 05:13:47 -0800 (PST)
Subject: [Freeswitch-users]  conference_auto_outcall_announce
Message-ID: <20955216.post@talk.nabble.com>


Hello,

I would like to call automatically a group of user and make them listening
an announcement before anybody talks. I have already added an extension in
the dialplan that worked. Actually, I use the extension 0911 in the default
dialplan of freeswitch which I have changed a little. This works fine. The
problem is to make the announcement.

1-
I have created a profile for this conference and add:
 < param name="enter-sound" value="conference/my_file.wav"/> 

It did not work. I have tested this, it seems to work only if someone enters
a conference which has been previously created. For instance after the 1st
participant has entered the conference room.


2- 
Then, I have seen there is the function "conference_auto_outcall_announce"
but I did not manage to make it work.
I have added in my extension:
<action application="set"
data="conference_auto_outcall_announce=play:path_file_to_play/file.wav"/>

I have got the following error:
[ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File
[/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
[System error : no such file or directory]

I have tried to change the path but something else is wrong.


3-
I have tried with the API command:
conference <confname> play <file_path>
It works from the console but not from the dialpan.


If someone could tell me where I am wrong or has another idea it would be
very helpfull.

Thanks,
Carole



-- 
View this message in context: http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20955216.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Thu Dec 11 05:57:52 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Dec 2008 07:57:52 -0600
Subject: [Freeswitch-users] conference_auto_outcall_announce
In-Reply-To: <20955216.post@talk.nabble.com>
References: <20955216.post@talk.nabble.com>
Message-ID: <0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>

Don't have play: in there and it should be fine.  Also if you want the  
absolute path you start it with /path/to/file.wav


/b

On Dec 11, 2008, at 7:13 AM, Carole O. wrote:

> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
> [System error : no such file or directory]



From john at loopfx.com  Thu Dec 11 06:29:58 2008
From: john at loopfx.com (John Rutherford)
Date: Thu, 11 Dec 2008 09:29:58 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local><87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local><49402977.4090808@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>
	<87f2f3b90812101452t7076bb6fo93f7a78bbfb0404f@mail.gmail.com>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360DB5@anniesue.loop.local>

No.  I wish it were that simple.  

I'm doing all of my testing on an internal network.  

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Wednesday, December 10, 2008 5:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

I smell a NAT... is there any NAT involved?

On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford <john at loopfx.com>
wrote:
> Okay. I just tried this.
>
>
>
> Now we're getting the audio going one way, but not the other.  So, I
can
> hear the person that I just transferred to, but they can't hear me.
>
>
>
> Anyone have any other ideas?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Raymond
> Chandler
> Sent: Wednesday, December 10, 2008 3:41 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> try blocking ICMP packets TO the MSS.... i had this exact same problem
a few
> months ago.... MSS starts sending RTP to FS before FS is ready to
accept....
> so the OS catches the port not open and returns an ICMP 3:3 back to
the
> MSS.... which in turn chokes on the queued up RTP and refuses to send
> anymore...
>
> -Ray
>
> John Rutherford wrote:
>
> I just emailed it to him.
>
>
>
> Thanks!
>
>
>
> -----Original Message-----
>
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>
> Michael Collins
>
> Sent: Wednesday, December 10, 2008 1:09 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
>
> wrote:
>
>
>
> I have a pcap, but I'm not able to see anything obviously wrong with
>
>
>
> it.
>
>
>
> We find that some equipment (in fact a lot of equipment) have features
>
> that cause issues to be quite non-obvious, so perhaps you could give
>
> the pcap to Brian for him to review. He's a total ace when it comes to
>
> bug hunting.
>
>
>
> -MC
>
>
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>
>
>
> Brian
>
>
>
> West
>
> Sent: Wednesday, December 10, 2008 12:59 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
>
>
>
>
> would be most helpful to capture a pcap of the entire thing by itself
>
>
>
> start
>
>
>
> to finish.
>
>
>
>
>
>
>
> /b
>
>
>
>
>
>
>
> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>
>
>
> No. I realize that's it's a B2BUA and that's exactly what we want.
>
>
>
>
>
>
>
> Everything with the transfer seems to work fine, except that there is
>
>
>
> no
>
>
>
> audio.
>
>
>
>
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From carole.olivier at enst.fr  Thu Dec 11 06:36:48 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 11 Dec 2008 06:36:48 -0800 (PST)
Subject: [Freeswitch-users] conference_auto_outcall_announce
In-Reply-To: <0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>
References: <20955216.post@talk.nabble.com>
	<0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>
Message-ID: <20956587.post@talk.nabble.com>


Hello,

Actually, I have already tried it but nothing happens: the file is not
played and there is no error. 
There is still a difference: if I configure it as you said, I can not be
listening anymore, there is simply nothing.

Would you have an idea? I have checked the path and the syntax 1 million
times so I do not think I make mistake there.

Thanks,
Carole



Brian West-3 wrote:
> 
> Don't have play: in there and it should be fine.  Also if you want the  
> absolute path you start it with /path/to/file.wav
> 
> 
> /b
> 
> On Dec 11, 2008, at 7:13 AM, Carole O. wrote:
> 
>> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
>> [System error : no such file or directory]
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20956587.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From intralanman at freeswitch.org  Thu Dec 11 06:39:11 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 11 Dec 2008 09:39:11 -0500
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360DB5@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local><87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local><49402977.4090808@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>	<87f2f3b90812101452t7076bb6fo93f7a78bbfb0404f@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360DB5@anniesue.loop.local>
Message-ID: <4941260F.90704@freeswitch.org>

can you send a pcap of sip and rtp with the new problem?

-Ray


John Rutherford wrote:
> No.  I wish it were that simple.  
>
> I'm doing all of my testing on an internal network.  
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
> Sent: Wednesday, December 10, 2008 5:53 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
> I smell a NAT... is there any NAT involved?
>
> On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford <john at loopfx.com>
> wrote:
>   
>> Okay. I just tried this.
>>
>>
>>
>> Now we're getting the audio going one way, but not the other.  So, I
>>     
> can
>   
>> hear the person that I just transferred to, but they can't hear me.
>>
>>
>>
>> Anyone have any other ideas?
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>     
> Raymond
>   
>> Chandler
>> Sent: Wednesday, December 10, 2008 3:41 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] No audio after transfer
>>
>>
>>
>> try blocking ICMP packets TO the MSS.... i had this exact same problem
>>     
> a few
>   
>> months ago.... MSS starts sending RTP to FS before FS is ready to
>>     
> accept....
>   
>> so the OS catches the port not open and returns an ICMP 3:3 back to
>>     
> the
>   
>> MSS.... which in turn chokes on the queued up RTP and refuses to send
>> anymore...
>>
>> -Ray
>>
>> John Rutherford wrote:
>>
>> I just emailed it to him.
>>
>>
>>
>> Thanks!
>>
>>
>>
>> -----Original Message-----
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>>
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>
>> Michael Collins
>>
>> Sent: Wednesday, December 10, 2008 1:09 PM
>>
>> To: freeswitch-users at lists.freeswitch.org
>>
>> Subject: Re: [Freeswitch-users] No audio after transfer
>>
>>
>>
>> On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
>>
>> wrote:
>>
>>
>>
>> I have a pcap, but I'm not able to see anything obviously wrong with
>>
>>
>>
>> it.
>>
>>
>>
>> We find that some equipment (in fact a lot of equipment) have features
>>
>> that cause issues to be quite non-obvious, so perhaps you could give
>>
>> the pcap to Brian for him to review. He's a total ace when it comes to
>>
>> bug hunting.
>>
>>
>>
>> -MC
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>>
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>
>>
>>
>> Brian
>>
>>
>>
>> West
>>
>> Sent: Wednesday, December 10, 2008 12:59 PM
>>
>> To: freeswitch-users at lists.freeswitch.org
>>
>> Subject: Re: [Freeswitch-users] No audio after transfer
>>
>>
>>
>>
>>
>>
>>
>> would be most helpful to capture a pcap of the entire thing by itself
>>
>>
>>
>> start
>>
>>
>>
>> to finish.
>>
>>
>>
>>
>>
>>
>>
>> /b
>>
>>
>>
>>
>>
>>
>>
>> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>>
>>
>>
>> No. I realize that's it's a B2BUA and that's exactly what we want.
>>
>>
>>
>>
>>
>>
>>
>> Everything with the transfer seems to work fine, except that there is
>>
>>
>>
>> no
>>
>>
>>
>> audio.
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>> Freeswitch-users mailing list
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>>
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>> Freeswitch-users mailing list
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>>
>> Freeswitch-users mailing list
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From intralanman at freeswitch.org  Thu Dec 11 06:55:30 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 11 Dec 2008 09:55:30 -0500
Subject: [Freeswitch-users] Configuring FreeSwitch
In-Reply-To: <1228987267.4843.6.camel@gathern.lan>
References: <1228987267.4843.6.camel@gathern.lan>
Message-ID: <494129E2.5010602@freeswitch.org>

Alexandru Nedelcu wrote:
> Hi,
>
> I'm a newbie trying to configure freeswitch.
>
> Our needs are simple: for starters we need to call external phones from
> freeswitch and play a wav file. 
>
> Now, we have the voice provider's IP (no user/password authentication),
> and we have the codec used (called "ulaw" in Asterisk, which I think is
> called PCMU?). And this is basically all that we specify in Asterisk.
>
>   
i think i answered all of this for you on irc yesterday....

use the bridge dialplan app to dial by ip similar to the following:
    <action application="bridge" 
data="sofia/${use_profile}/number at ip.address"/>

http://wiki.freeswitch.org/wiki/Sofia#Syntax might also help you out a 
little



From alex at sinapticode.ro  Thu Dec 11 07:25:01 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 17:25:01 +0200
Subject: [Freeswitch-users] Having trouble with capturing key presses from
	Javascript
Message-ID: <1229009101.4843.18.camel@gathern.lan>

Hi,

I've tried a simple example from here:
http://wiki.freeswitch.org/wiki/Session_streamFile
You can view my code here:
http://paste.scsys.co.uk/paste

The problem is, streamFile doesn't fire the event on keypress.
Is there anything wrong with my code?

Thank you,

-- 
Alexandru Nedelcu
Software Developer, Sinapticode



From alex at sinapticode.ro  Thu Dec 11 07:59:14 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 17:59:14 +0200
Subject: [Freeswitch-users] Having trouble with capturing key presses
	from Javascript
In-Reply-To: <1229009101.4843.18.camel@gathern.lan>
References: <1229009101.4843.18.camel@gathern.lan>
Message-ID: <1229011154.4843.21.camel@gathern.lan>

On Thu, 2008-12-11 at 17:25 +0200, Alexandru Nedelcu wrote:
> Hi,
> 
> I've tried a simple example from here:
> http://wiki.freeswitch.org/wiki/Session_streamFile
> You can view my code here:
> http://paste.scsys.co.uk/paste

Sorry, the correct link was:
http://scsys.co.uk:8001/21364

> 
> The problem is, streamFile doesn't fire the event on keypress.
> Is there anything wrong with my code?

Another thing: in Asterisk I had the following setting in sip.conf ...
dtmfmode=inband
Without it key presses wasn't working in Asterisk.

Is there something similar for Freeswitch?



From intralanman at freeswitch.org  Thu Dec 11 08:22:11 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 11 Dec 2008 11:22:11 -0500
Subject: [Freeswitch-users] Having trouble with capturing key presses
 from Javascript
In-Reply-To: <1229011154.4843.21.camel@gathern.lan>
References: <1229009101.4843.18.camel@gathern.lan>
	<1229011154.4843.21.camel@gathern.lan>
Message-ID: <49413E33.2030604@freeswitch.org>

use the start_dtmf app to get inband dtmf

-Ray

Alexandru Nedelcu wrote:
> On Thu, 2008-12-11 at 17:25 +0200, Alexandru Nedelcu wrote:
>   
>> Hi,
>>
>> I've tried a simple example from here:
>> http://wiki.freeswitch.org/wiki/Session_streamFile
>> You can view my code here:
>> http://paste.scsys.co.uk/paste
>>     
>
> Sorry, the correct link was:
> http://scsys.co.uk:8001/21364
>
>   
>> The problem is, streamFile doesn't fire the event on keypress.
>> Is there anything wrong with my code?
>>     
>
> Another thing: in Asterisk I had the following setting in sip.conf ...
> dtmfmode=inband
> Without it key presses wasn't working in Asterisk.
>
> Is there something similar for Freeswitch?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From alex at sinapticode.ro  Thu Dec 11 08:29:35 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 18:29:35 +0200
Subject: [Freeswitch-users] Having trouble with capturing key
	presses	from Javascript
In-Reply-To: <49413E33.2030604@freeswitch.org>
References: <1229009101.4843.18.camel@gathern.lan>
	<1229011154.4843.21.camel@gathern.lan>
	<49413E33.2030604@freeswitch.org>
Message-ID: <1229012975.4843.23.camel@gathern.lan>

Can the app be initiated from JS?


On Thu, 2008-12-11 at 11:22 -0500, Raymond Chandler wrote:
> use the start_dtmf app to get inband dtmf
> 
> -Ray
> 
> Alexandru Nedelcu wrote: 
> > On Thu, 2008-12-11 at 17:25 +0200, Alexandru Nedelcu wrote:
> >   
> > > Hi,
> > > 
> > > I've tried a simple example from here:
> > > http://wiki.freeswitch.org/wiki/Session_streamFile
> > > You can view my code here:
> > > http://paste.scsys.co.uk/paste
> > >     
> > 
> > Sorry, the correct link was:
> > http://scsys.co.uk:8001/21364
> > 
> >   
> > > The problem is, streamFile doesn't fire the event on keypress.
> > > Is there anything wrong with my code?
> > >     
> > 
> > Another thing: in Asterisk I had the following setting in sip.conf ...
> > dtmfmode=inband
> > Without it key presses wasn't working in Asterisk.
> > 
> > Is there something similar for Freeswitch?
> > 
> > 
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >   
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Dec 11 08:39:48 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Dec 2008 10:39:48 -0600
Subject: [Freeswitch-users] Having trouble with capturing key
	presses	from Javascript
In-Reply-To: <1229012975.4843.23.camel@gathern.lan>
References: <1229009101.4843.18.camel@gathern.lan>
	<1229011154.4843.21.camel@gathern.lan>
	<49413E33.2030604@freeswitch.org>
	<1229012975.4843.23.camel@gathern.lan>
Message-ID: <039F34E7-5CFF-4185-A29A-FB62BC6A28AE@freeswitch.org>

session.execute("start_dtmf")

/b

On Dec 11, 2008, at 10:29 AM, Alexandru Nedelcu wrote:

> Can the app be initiated from JS?



From john at loopfx.com  Thu Dec 11 10:42:54 2008
From: john at loopfx.com (John Rutherford)
Date: Thu, 11 Dec 2008 13:42:54 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local><87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local><49402977.4090808@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>	<87f2f3b90812101452t7076bb6fo93f7a78bbfb0404f@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360DB5@anniesue.loop.local>
	<4941260F.90704@freeswitch.org>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360E73@anniesue.loop.local>

Sent.  

 

Let me know if you see anything. I'm not able to see anything wrong.

 

Thanks,

John

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Raymond Chandler
Sent: Thursday, December 11, 2008 9:39 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

can you send a pcap of sip and rtp with the new problem?

-Ray 


John Rutherford wrote: 

No.  I wish it were that simple.  
 
I'm doing all of my testing on an internal network.  
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Wednesday, December 10, 2008 5:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer
 
I smell a NAT... is there any NAT involved?
 
On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford <john at loopfx.com>
<mailto:john at loopfx.com> 
wrote:
  

	Okay. I just tried this.
	 
	 
	 
	Now we're getting the audio going one way, but not the other.
So, I
	    

can
  

	hear the person that I just transferred to, but they can't hear
me.
	 
	 
	 
	Anyone have any other ideas?
	 
	 
	 
	From: freeswitch-users-bounces at lists.freeswitch.org
	[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
	    

Raymond
  

	Chandler
	Sent: Wednesday, December 10, 2008 3:41 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] No audio after transfer
	 
	 
	 
	try blocking ICMP packets TO the MSS.... i had this exact same
problem
	    

a few
  

	months ago.... MSS starts sending RTP to FS before FS is ready
to
	    

accept....
  

	so the OS catches the port not open and returns an ICMP 3:3 back
to
	    

the
  

	MSS.... which in turn chokes on the queued up RTP and refuses to
send
	anymore...
	 
	-Ray
	 
	John Rutherford wrote:
	 
	I just emailed it to him.
	 
	 
	 
	Thanks!
	 
	 
	 
	-----Original Message-----
	 
	From: freeswitch-users-bounces at lists.freeswitch.org
	 
	[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
	 
	Michael Collins
	 
	Sent: Wednesday, December 10, 2008 1:09 PM
	 
	To: freeswitch-users at lists.freeswitch.org
	 
	Subject: Re: [Freeswitch-users] No audio after transfer
	 
	 
	 
	On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford
<john at loopfx.com> <mailto:john at loopfx.com> 
	 
	wrote:
	 
	 
	 
	I have a pcap, but I'm not able to see anything obviously wrong
with
	 
	 
	 
	it.
	 
	 
	 
	We find that some equipment (in fact a lot of equipment) have
features
	 
	that cause issues to be quite non-obvious, so perhaps you could
give
	 
	the pcap to Brian for him to review. He's a total ace when it
comes to
	 
	bug hunting.
	 
	 
	 
	-MC
	 
	 
	 
	 
	 
	 
	 
	 
	 
	From: freeswitch-users-bounces at lists.freeswitch.org
	 
	[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
	 
	 
	 
	Brian
	 
	 
	 
	West
	 
	Sent: Wednesday, December 10, 2008 12:59 PM
	 
	To: freeswitch-users at lists.freeswitch.org
	 
	Subject: Re: [Freeswitch-users] No audio after transfer
	 
	 
	 
	 
	 
	 
	 
	would be most helpful to capture a pcap of the entire thing by
itself
	 
	 
	 
	start
	 
	 
	 
	to finish.
	 
	 
	 
	 
	 
	 
	 
	/b
	 
	 
	 
	 
	 
	 
	 
	On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
	 
	 
	 
	No. I realize that's it's a B2BUA and that's exactly what we
want.
	 
	 
	 
	 
	 
	 
	 
	Everything with the transfer seems to work fine, except that
there is
	 
	 
	 
	no
	 
	 
	 
	audio.
	 
	 
	 
	 
	 
	 
	 
	_______________________________________________
	 
	Freeswitch-users mailing list
	 
	Freeswitch-users at lists.freeswitch.org
	 
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	 
	 
	 
	 
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	 
	 
	http://www.freeswitch.org
	 
	 
	 
	 
	 
	 
	 
	 
	 
	_______________________________________________
	 
	Freeswitch-users mailing list
	 
	Freeswitch-users at lists.freeswitch.org
	 
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	http://www.freeswitch.org
	 
	 
	 
	_______________________________________________
	 
	Freeswitch-users mailing list
	 
	Freeswitch-users at lists.freeswitch.org
	 
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	http://www.freeswitch.org
	 
	 
	 
	 
	 
	_______________________________________________
	Freeswitch-users mailing list
	Freeswitch-users at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	http://www.freeswitch.org
	 
	 
	    

 
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

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From vkobashi at ydeasolutions.com.br  Thu Dec 11 11:59:31 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Thu, 11 Dec 2008 16:59:31 -0300
Subject: [Freeswitch-users]  LDAP Integration
Message-ID: <49417123.10709@ydeasolutions.com.br>

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From vkobashi at ydeasolutions.com.br  Thu Dec 11 12:16:56 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Thu, 11 Dec 2008 17:16:56 -0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49417123.10709@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>
Message-ID: <49417538.9040203@ydeasolutions.com.br>

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From hads at nice.net.nz  Thu Dec 11 11:42:00 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Fri, 12 Dec 2008 08:42:00 +1300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49417538.9040203@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>
	<49417538.9040203@ydeasolutions.com.br>
Message-ID: <200812120842.00808.hads@nice.net.nz>

On Friday 12 December 2008 09:16:56 Vinicius Kobashi wrote:
> i found another module called mod_xml_curl and loaded it to freeswitch
> too... but still it shows me the following error:
>
>  2008-12-11 17:04:04 [WARNING] sofia_reg.c:1501 sofia_reg_parse_auth()
> Can't find user [username at freeswitchserver.com] You must define a domain
> called 'freeswitchserver.com' in your directory and add a user with the
> id="username" attribute and you must configure your device to use the
> proper domain in it's authentication credentials.
>
>  does anyone got an idea?

Yes, you need to define a domain called 'freeswitchserver.com' in your 
directory and add a user with the id="username" just like the error message 
says.

The directory files are in conf/directory/

If you would like to read up on mod_xml_curl there is a detailed page on the 
wiki;

http://wiki.freeswitch.org/wiki/Mod_xml_curl

hads
-- 
http://nicegear.co.nz
VoIP, DVB and other Linux compatible hardware.


From vkobashi at ydeasolutions.com.br  Thu Dec 11 13:35:12 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Thu, 11 Dec 2008 18:35:12 -0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <200812120842.00808.hads@nice.net.nz>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>
	<200812120842.00808.hads@nice.net.nz>
Message-ID: <49418790.60001@ydeasolutions.com.br>

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From brian at freeswitch.org  Thu Dec 11 12:40:18 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Dec 2008 14:40:18 -0600
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49418790.60001@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>
	<200812120842.00808.hads@nice.net.nz>
	<49418790.60001@ydeasolutions.com.br>
Message-ID: <2F2BD503-9199-4131-998A-3777809624CD@freeswitch.org>

Thats already fixed too.. update.

/b

On Dec 11, 2008, at 3:35 PM, Vinicius Kobashi wrote:

> ok ill try that
>
> i found another module thats mod_xml_ldap
>
> but when i try to load it, during compiling i get the 404 error http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz 
>  file not found....
> ill try to download it myself and then try to compile freeswitch  
> again and test
>
> =D thankz for the fast answer

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From msc at freeswitch.org  Thu Dec 11 12:41:56 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Dec 2008 12:41:56 -0800
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49418790.60001@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>
	<49417538.9040203@ydeasolutions.com.br>
	<200812120842.00808.hads@nice.net.nz>
	<49418790.60001@ydeasolutions.com.br>
Message-ID: <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>

Please confirm your svn rev - I believe this was fixed recently. Do
"make current" in your source directory.
-MC

On Thu, Dec 11, 2008 at 1:35 PM, Vinicius Kobashi
<vkobashi at ydeasolutions.com.br> wrote:
> ok ill try that
>
> i found another module thats mod_xml_ldap
>
> but when i try to load it, during compiling i get the 404 error
> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz file not
> found....
> ill try to download it myself and then try to compile freeswitch again and
> test
>
> =D thankz for the fast answer
>
> Hadley Rich escreveu:
>
> On Friday 12 December 2008 09:16:56 Vinicius Kobashi wrote:
>
>
> i found another module called mod_xml_curl and loaded it to freeswitch
> too... but still it shows me the following error:
>
>  2008-12-11 17:04:04 [WARNING] sofia_reg.c:1501 sofia_reg_parse_auth()
> Can't find user [username at freeswitchserver.com] You must define a domain
> called 'freeswitchserver.com' in your directory and add a user with the
> id="username" attribute and you must configure your device to use the
> proper domain in it's authentication credentials.
>
>  does anyone got an idea?
>
>
> Yes, you need to define a domain called 'freeswitchserver.com' in your
> directory and add a user with the id="username" just like the error message
> says.
>
> The directory files are in conf/directory/
>
> If you would like to read up on mod_xml_curl there is a detailed page on the
> wiki;
>
> http://wiki.freeswitch.org/wiki/Mod_xml_curl
>
> hads
>
>
> --
>
>
> Vinicius Kobashi
> Infra-Estrutura
>
> Ydea Desenvolvimento de Software LTDA.
> Av. Adolfo Pinheiro, 2338 - Alto da Boa Vista
> CEP.:04734-004 - S?o Paulo - SP
> Tel.: 55-11-5523-0333
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From erick at junctionnetworks.com  Thu Dec 11 12:52:38 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Thu, 11 Dec 2008 15:52:38 -0500
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
References: 493ED9E6.3000803@junctionnetworks.com
Message-ID: <49417D96.1090805@junctionnetworks.com>

Thanks Dave,

Actually I realized my problem (stupid mistake of course).  For anyone else
trying to use the fs_path variable the value needs to be a fully 
qualified SIP
URI, e.g. "bob at bar.com;fs_path=sip:host.domain.net", notice it being 
prefaced
with the "sip:", my problem was that  I was only entering
the host name.  Then somewhere down in mod_sofia it must have decided that
it didn't like that and just closed the channel.

Hope this helps somebody who gets stuck like I did.

Cheers,

Erick

> Hi Erick,
>
> Not sure if you've tried this (or if it'll help), but you can force
> routing in the dialplan like so:
> <action application="set" data="sip_h_Route=<sip:@11.22.33.44;lr>" />
> <action application="bridge" data="sofia/gateway/gw/$1"/>
>
> Cheers --
>
> Dave



From msc at freeswitch.org  Thu Dec 11 12:59:46 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Dec 2008 12:59:46 -0800
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <49417D96.1090805@junctionnetworks.com>
References: <49417D96.1090805@junctionnetworks.com>
Message-ID: <87f2f3b90812111259ob0a78bbw747afdc52251c2cb@mail.gmail.com>

On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson
<erick at junctionnetworks.com> wrote:
> Thanks Dave,
>
> Actually I realized my problem (stupid mistake of course).  For anyone else
> trying to use the fs_path variable the value needs to be a fully
> qualified SIP
> URI, e.g. "bob at bar.com;fs_path=sip:host.domain.net", notice it being
> prefaced
> with the "sip:", my problem was that  I was only entering
> the host name.  Then somewhere down in mod_sofia it must have decided that
> it didn't like that and just closed the channel.

Erick, thanks for the clarification! I'll get it put on the wiki right away.
-MC

>
> Hope this helps somebody who gets stuck like I did.
>
> Cheers,
>
> Erick
>
>> Hi Erick,
>>
>> Not sure if you've tried this (or if it'll help), but you can force
>> routing in the dialplan like so:
>> <action application="set" data="sip_h_Route=<sip:@11.22.33.44;lr>" />
>> <action application="bridge" data="sofia/gateway/gw/$1"/>
>>
>> Cheers --
>>
>> Dave
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From vkobashi at ydeasolutions.com.br  Thu Dec 11 14:49:23 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Thu, 11 Dec 2008 19:49:23 -0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>
	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>
Message-ID: <494198F3.10806@ydeasolutions.com.br>

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From erick at junctionnetworks.com  Thu Dec 11 13:08:52 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Thu, 11 Dec 2008 16:08:52 -0500
Subject: [Freeswitch-users]  Restricting SIP methods in Allow header
Message-ID: <49418164.7080206@junctionnetworks.com>

Is it possible it configure a sip-profile so that the UA reports a 
restricted
set of methods in the SIP allow header?  For instance, I would like to 
remove
all methods except  ACK, BYE, CANCEL, NOTIFY, and PRACK

Thanks

Erick

-- 
Erick Johnson


From anthony.minessale at gmail.com  Thu Dec 11 14:36:20 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Dec 2008 16:36:20 -0600
Subject: [Freeswitch-users] Restricting SIP methods in Allow header
In-Reply-To: <49418164.7080206@junctionnetworks.com>
References: <49418164.7080206@junctionnetworks.com>
Message-ID: <191c3a030812111436v2972290bsf889ce2b5db4488@mail.gmail.com>

no but there are many options in the profile that directly control what
methods we advertise based on if they are enabled.


On Thu, Dec 11, 2008 at 3:08 PM, Erick Johnson
<erick at junctionnetworks.com>wrote:

> Is it possible it configure a sip-profile so that the UA reports a
> restricted
> set of methods in the SIP allow header?  For instance, I would like to
> remove
> all methods except  ACK, BYE, CANCEL, NOTIFY, and PRACK
>
> Thanks
>
> Erick
>
> --
> Erick Johnson
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Dec 11 15:39:26 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Dec 2008 17:39:26 -0600
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <1228978862.31447.5.camel@quentusrex-desktop>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
	<1228978862.31447.5.camel@quentusrex-desktop>
Message-ID: <191c3a030812111539k5b181c96qbfa13f5015c8bb98@mail.gmail.com>

issue should be fixed in SVN r10723

On Thu, Dec 11, 2008 at 1:01 AM, William King <william at channelxstream.com>wrote:

> Carole,
>
> 1.0.1 is known to be broken now. Can you go into the directory where you
> have the freeswitch source from the svn repo and type 'make current'.
> This will update freeswitch to trunk.
>
> -William
>
> On Wed, 2008-12-10 at 22:51 -0800, Carole O. wrote:
> > Hello,
> >
> > I am running the version 1.0.1.
> > Do you still need me to run the debugging?
> >
> > Carole
> >
> >
> > Brian West-3 wrote:
> > >
> > > I have already labbed this up on SVN trunk and I don't get a segfault
> > > but I get something else that prevents it from working properly.    We
> > > are working on it today.  Also what version are you running?
> > >
> > > /b
> > >
> > > On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
> > >
> > >> Thanks for reporting this. It would be helpful to know a bit more. Can
> > >> you start freeswitch and press F12 (or type "version" at the CLI) and
> > >> report back what it says?
> > >> Also, a backtrace (bt) is generally useful. If you could produce a
> > >> "bt" and a "bt full" from you core file that would be extremely
> > >> helpful.
> > >>
> > >> see this link for more information:
> > >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
> > >>
> > >> you should have a "core" file for each segfault that occurred. Use the
> > >> gdb program to get the back trace:
> > >>
> > >> gdb /path/to/fs/binary core.xxx
> > >>
> > >> then capture the output from these two commands:
> > >>
> > >> bt <enter>
> > >> bt full <enter>
> > >>
> > >> When you type those commands you'll see tons of debugging info;
> > >> capture that and put it in a pastebin (pastebin.freeswitch.org) then
> > >> report back here.
> > >>
> > >> You can exit the gdb debugger by typing q <enter>
> > >>
> > >> Thanks for helping us collect information!
> > >>
> > >> -MC
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> --
> William King
> Cell: 253-686-5518
> E-mail: william at channelxstream.com
> Channel XStream
> www.channelxstream.com
> 1-877-600-6786
>
> If there is a possibility that any information in our conversation might
> be considered 'private' or 'sensitive' such as passwords, account
> information, legal or financial information, or anything else that you
> would consider 'private' or 'sensitive' communications. It is better to
> always err on the side of security. Please encrypt the e-mail using
> my gpg key: 95C9D5B3.
>
> If you are unfamiliar with e-mail encryption feel free to let me know
> and I can help you establish the proper protocols and procedures.
> https://help.ubuntu.com/community/GnuPrivacyGuardHowto
>
> Get my gpg key: gpg --recv-key --keyserver keyserver.ubuntu.com 95C9D5B3
> Key Fingerprint: EA6F B2EE 1846 55D4 FFD9  80BA 6489 B48C 95C9 D5B3
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From dalechase at yahoo.com  Thu Dec 11 16:51:30 2008
From: dalechase at yahoo.com (dalechase at yahoo.com)
Date: Thu, 11 Dec 2008 16:51:30 -0800 (PST)
Subject: [Freeswitch-users] config help: openzap and T1 A102u
Message-ID: <603978.74977.qm@web36108.mail.mud.yahoo.com>

I am stuck trying to bring up freeswitch with openzap on a Sangoma A102u T1 card.
Works fine with asterisk.

Please point out where I am failing to configure properly.

Running Linux version 2.6.9-34.ELsmp on a Dell Celeron

% wanrouter hwprobe verbose

-----------------------------------------
| Wanpipe Hardware Probe Info (verbose) |
-----------------------------------------
1 . AFT-A102u  : SLOT=1 : BUS=2 : IRQ=145 : CPU=A : PORT=PRI : V=25
+01:PMC4351:PCI
2 . AFT-A102u  : SLOT=1 : BUS=2 : IRQ=145 : CPU=B : PORT=PRI : V=25
+01:PMC4351:PCI

Card Cnt: S508=0  S514X=0  S518=0  A101-2=1  A104=0  A300=0  A200=0  A108=0

% cat /usr/local/freeswitch/conf/autoload_configs/open
openmrcp.conf.xml  openzap.conf.xml
[root at pbxtra1466 freeswitch]# cat /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <!-- one entry here per openzap span -->
  <analog_spans>
    <span id="1">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <!--<param name="enable-analog-option" value="call-swap"/>-->
      <!--<param name="enable-analog-option" value="3-way"/>-->
      <param name="tonegroup" value="us"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
    <span id="2">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <!--<param name="enable-analog-option" value="call-swap"/>-->
      <!--<param name="enable-analog-option" value="3-way"/>-->
      <param name="tonegroup" value="us"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </analog_spans>

</configuration>

% cat /etc/openzap/openzap.conf
[span wanpipe]
trunk_type => t1
b-channel => 1:1-23
d-channel=> 1:24

[span wanpipe]
trunk_type => t1
b-channel => 2:25-47
d-channel=> 2:48

% cat /etc/openzap/wanpipe.conf
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750

% cat /etc/wanpipe/wanpipe1.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Tue Dec 12 16:21:45 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/sbin/wancfg program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe1 = WAN_AFT, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE       = AFT
S514CPU         = A
CommPort        = PRI
AUTO_PCISLOT    = NO
PCISLOT         = 1
PCIBUS          = 2
FE_MEDIA        = T1
FE_LCODE        = B8ZS
FE_FRAME        = ESF
FE_LINE         = 1
TE_CLOCK        = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE        = NO
LBO             = 0DB
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END = NO
TDMV_SPAN       = 1
TDMV_DCHAN      = 24

[w1g1]
ACTIVE_CH       = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC       = NO

% cat /etc/wanpipe/wanpipe2.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Tue Dec 12 16:21:45 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/sbin/wancfg program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe2 = WAN_AFT, Comment

[interfaces]
w2g1 = wanpipe2, , TDM_VOICE, Comment

[wanpipe2]
CARD_TYPE       = AFT
S514CPU         = B
CommPort        = PRI
AUTO_PCISLOT    = NO
PCISLOT         = 1
PCIBUS          = 2
FE_MEDIA        = T1
FE_LCODE        = B8ZS
FE_FRAME        = ESF
FE_LINE         = 1
TE_CLOCK        = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE        = NO
LBO             = 0DB
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END = NO
TDMV_SPAN       = 2
TDMV_DCHAN      = 24

[w2g1]
ACTIVE_CH       = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC       = NO

freeswitch at hostname-elided> load mod_openzap
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c1
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c2
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c3
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c4
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c5
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c6
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c7
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c8
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c9
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c10
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c11
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c12
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c13
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c14
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c15
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c16
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c17
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c18
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c19
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c20
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c21
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c22
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c23
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c24
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c25
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c26
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c27
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c28
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c29
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c30
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c31
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c32
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c33
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c34
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c35
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c36
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c37
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c38
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c39
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c40
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c41
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c42
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c43
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c44
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c45
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c46
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c47
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c48
2008-12-11 16:23:08 [INFO] zap_io.c:2068 load_config() Configured 0 channel(s)
2008-12-11 16:23:08 [ERR] zap_io.c:2161 zap_global_init() No modules configured!
2008-12-11 16:23:08 [ERR] mod_openzap.c:1861 mod_openzap_load() Error loading OpenZAP
2008-12-11 16:23:08 [CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so
**Module load routine returned an error**


      


From zolotov at altron.ua  Fri Dec 12 01:00:01 2008
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Fri, 12 Dec 2008 11:00:01 +0200
Subject: [Freeswitch-users] config help: openzap and T1 A102u
References: <603978.74977.qm@web36108.mail.mud.yahoo.com>
Message-ID: <004001c95c38$04da0210$6d02a8c0@opos20>

  Did you try ./wanrouter start before starting FreeSWITCH ?
----- Original Message ----- 
From: <dalechase at yahoo.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, December 12, 2008 2:51 AM
Subject: [Freeswitch-users] config help: openzap and T1 A102u


>I am stuck trying to bring up freeswitch with openzap on a Sangoma A102u T1 
>card.
> Works fine with asterisk.
>
> Please point out where I am failing to configure properly.
>
> Running Linux version 2.6.9-34.ELsmp on a Dell Celeron
>
> % wanrouter hwprobe verbose
>
> -----------------------------------------
> | Wanpipe Hardware Probe Info (verbose) |
> -----------------------------------------
> 1 . AFT-A102u  : SLOT=1 : BUS=2 : IRQ=145 : CPU=A : PORT=PRI : V=25
> +01:PMC4351:PCI
> 2 . AFT-A102u  : SLOT=1 : BUS=2 : IRQ=145 : CPU=B : PORT=PRI : V=25
> +01:PMC4351:PCI
>
> Card Cnt: S508=0  S514X=0  S518=0  A101-2=1  A104=0  A300=0  A200=0 
> A108=0
>
> % cat /usr/local/freeswitch/conf/autoload_configs/open
> openmrcp.conf.xml  openzap.conf.xml
> [root at pbxtra1466 freeswitch]# cat 
> /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="0"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>  <!-- one entry here per openzap span -->
>  <analog_spans>
>    <span id="1">
>      <!--<param name="hold-music" value="$${moh_uri}"/>-->
>      <!--<param name="enable-analog-option" value="call-swap"/>-->
>      <!--<param name="enable-analog-option" value="3-way"/>-->
>      <param name="tonegroup" value="us"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>      <!-- regex to stop dialing when it matches -->
>      <!--<param name="dial-regex" value="5555"/>-->
>      <!-- regex to stop dialing when it does not match -->
>      <!--<param name="fail-dial-regex" value="^5"/>-->
>    </span>
>    <span id="2">
>      <!--<param name="hold-music" value="$${moh_uri}"/>-->
>      <!--<param name="enable-analog-option" value="call-swap"/>-->
>      <!--<param name="enable-analog-option" value="3-way"/>-->
>      <param name="tonegroup" value="us"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>      <!-- regex to stop dialing when it matches -->
>      <!--<param name="dial-regex" value="5555"/>-->
>      <!-- regex to stop dialing when it does not match -->
>      <!--<param name="fail-dial-regex" value="^5"/>-->
>    </span>
>  </analog_spans>
>
> </configuration>
>
> % cat /etc/openzap/openzap.conf
> [span wanpipe]
> trunk_type => t1
> b-channel => 1:1-23
> d-channel=> 1:24
>
> [span wanpipe]
> trunk_type => t1
> b-channel => 2:25-47
> d-channel=> 2:48
>
> % cat /etc/openzap/wanpipe.conf
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
>
> % cat /etc/wanpipe/wanpipe1.conf
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Date: Tue Dec 12 16:21:45 UTC 2006
> #
> # Note: This file was generated automatically
> #       by /usr/sbin/wancfg program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
> [devices]
> wanpipe1 = WAN_AFT, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE, Comment
>
> [wanpipe1]
> CARD_TYPE       = AFT
> S514CPU         = A
> CommPort        = PRI
> AUTO_PCISLOT    = NO
> PCISLOT         = 1
> PCIBUS          = 2
> FE_MEDIA        = T1
> FE_LCODE        = B8ZS
> FE_FRAME        = ESF
> FE_LINE         = 1
> TE_CLOCK        = NORMAL
> TE_REF_CLOCK    = 0
> TE_SIG_MODE     = CCS
> TE_HIGHIMPEDANCE        = NO
> LBO             = 0DB
> FE_TXTRISTATE   = NO
> MTU             = 1500
> UDPPORT         = 9000
> TTL             = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN       = 1
> TDMV_DCHAN      = 24
>
> [w1g1]
> ACTIVE_CH       = ALL
> TDMV_ECHO_OFF   = NO
> TDMV_HWEC       = NO
>
> % cat /etc/wanpipe/wanpipe2.conf
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Date: Tue Dec 12 16:21:45 UTC 2006
> #
> # Note: This file was generated automatically
> #       by /usr/sbin/wancfg program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
> [devices]
> wanpipe2 = WAN_AFT, Comment
>
> [interfaces]
> w2g1 = wanpipe2, , TDM_VOICE, Comment
>
> [wanpipe2]
> CARD_TYPE       = AFT
> S514CPU         = B
> CommPort        = PRI
> AUTO_PCISLOT    = NO
> PCISLOT         = 1
> PCIBUS          = 2
> FE_MEDIA        = T1
> FE_LCODE        = B8ZS
> FE_FRAME        = ESF
> FE_LINE         = 1
> TE_CLOCK        = NORMAL
> TE_REF_CLOCK    = 0
> TE_SIG_MODE     = CCS
> TE_HIGHIMPEDANCE        = NO
> LBO             = 0DB
> FE_TXTRISTATE   = NO
> MTU             = 1500
> UDPPORT         = 9000
> TTL             = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN       = 2
> TDMV_DCHAN      = 24
>
> [w2g1]
> ACTIVE_CH       = ALL
> TDMV_ECHO_OFF   = NO
> TDMV_HWEC       = NO
>
> freeswitch at hostname-elided> load mod_openzap
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c1
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c2
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c3
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c4
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c5
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c6
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c7
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c8
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c9
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c10
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c11
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c12
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c13
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c14
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c15
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c16
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c17
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c18
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c19
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c20
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c21
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c22
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c23
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c24
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c25
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c26
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c27
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c28
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c29
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c30
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c31
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c32
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c33
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c34
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c35
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c36
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c37
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c38
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c39
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c40
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c41
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c42
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c43
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c44
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c45
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c46
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c47
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c48
> 2008-12-11 16:23:08 [INFO] zap_io.c:2068 load_config() Configured 0 
> channel(s)
> 2008-12-11 16:23:08 [ERR] zap_io.c:2161 zap_global_init() No modules 
> configured!
> 2008-12-11 16:23:08 [ERR] mod_openzap.c:1861 mod_openzap_load() Error 
> loading OpenZAP
> 2008-12-11 16:23:08 [CRIT] switch_loadable_module.c:756 
> switch_loadable_module_load_file() Error Loading module 
> /usr/local/freeswitch/mod/mod_openzap.so
> **Module load routine returned an error**
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From alex at sinapticode.ro  Fri Dec 12 03:38:47 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 13:38:47 +0200
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers
Message-ID: <1229081927.4100.7.camel@gathern.lan>

Hi,

I'm working on a simple dialer, and I have the following problem: the
audio file starts playing before the user answeres the phone (while it's
ringing). It only works when I introduce a delay, but that doesn't seem
right.

For instance in the asterisk context referred in the call files, I had:

exten => s,4,Answer
exten => s,n,Wait(2)
exten => s,n,Background(${SOUNDFILE})
And indeed it played a soundfile 2 seconds after the called person
picked up the phone

In FS I currently initiate calls like this:

session.waitForAnswer(10000);

if (session.ready()) {
    session.sleep(2000);
    session.streamFile(/*...*/);
}

Is this right?



From alex at sinapticode.ro  Fri Dec 12 04:10:29 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 14:10:29 +0200
Subject: [Freeswitch-users] Freeswitch logging
Message-ID: <1229083829.4100.11.camel@gathern.lan>

Hi,

I see that mod_cdr is marked as being non-functional on the wiki. I'm working on a dialer and I need a way to log information about calls.

What module should I use?

Thanks,



From hads at nice.net.nz  Fri Dec 12 04:26:56 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Sat, 13 Dec 2008 01:26:56 +1300
Subject: [Freeswitch-users] Freeswitch logging
In-Reply-To: <1229083829.4100.11.camel@gathern.lan>
References: <1229083829.4100.11.camel@gathern.lan>
Message-ID: <200812130126.56386.hads@nice.net.nz>

On Saturday 13 December 2008 01:10:29 Alexandru Nedelcu wrote:
> Hi,
>
> I see that mod_cdr is marked as being non-functional on the wiki. I'm
> working on a dialer and I need a way to log information about calls.
>
> What module should I use?

This was answered on IRC and a note added to the mod_cdr wiki page.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier


From jbr at consiglia.dk  Fri Dec 12 05:30:27 2008
From: jbr at consiglia.dk (Jon Bruel)
Date: Fri, 12 Dec 2008 14:30:27 +0100
Subject: [Freeswitch-users] fifo.conf.xml usage
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE027E3B3E6E@SBS2008SERVER.consiglia.local>

I'm happy to see that you can add consumers to queues using the fifo.conf.xml configuration file. I have made some tests and I hope it may lead to a more universal way of setting up queues for small organisations than the one I have described in the wiki, and which includes (too) many javascripts. I have some questions to clarify my understanding. Using the fifo.conf.xml, I find:
 - That the consumers continue to ring after the caller has abandoned the queue. Is there a way to avoid this?
Further:
 - Is there a way to control the caller_id_name/number presented to the consumer?
 - Is there a way to control the ringing tone in the consumers like the one which can be used in the dialplan?
 - Can the fifo.conf.xml refer to an ODBC connection in order to get the members from a database?
Finally, thanks for all the good work everybody in the FS community has put into FS, I truly believe in the possibilities of this product. Checking the hits on Google certainly indicates you moving into the right direction. /Jon

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From d at d-man.org  Fri Dec 12 05:48:14 2008
From: d at d-man.org (Darren Schreiber)
Date: Fri, 12 Dec 2008 05:48:14 -0800
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers
In-Reply-To: <1229081927.4100.7.camel@gathern.lan>
References: <1229081927.4100.7.camel@gathern.lan>
Message-ID: <05FEA4243A6C422DB5A3D7838AA709FE@test>

How are you originating calls? You probably need to add
{ignore_early_media=true}. This tells FreeSWITCH not to return from
origination when early media (progress/ringing) was received (I think
anyway)...

See http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media

There is a sample of this in use with the originate command here:
http://wiki.freeswitch.org/wiki/Mod_commands#originate (about halfway down)

Setting channel variables before doing the originate 

   originate {ignore_early_media=true}sofia/mydomain.com/18005551212 at 1.2.3.4
15555551212



Since you are making a dialer, you may want to start the originations in the
background and move on to the next call while tweaking the timeout value for
originated calls. From the WIKI again:

"You can originate a call in the background (asynchronously) and playback a
message with a 60 second timeout. 

   bgapi originate
{ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number
&playback(message)"

- Darren

 

-----Original Message-----
From: Alexandru Nedelcu [mailto:alex at sinapticode.ro] 
Sent: Friday, December 12, 2008 3:39 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers

Hi,

I'm working on a simple dialer, and I have the following problem: the audio
file starts playing before the user answeres the phone (while it's ringing).
It only works when I introduce a delay, but that doesn't seem right.

For instance in the asterisk context referred in the call files, I had:

exten => s,4,Answer
exten => s,n,Wait(2)
exten => s,n,Background(${SOUNDFILE})
And indeed it played a soundfile 2 seconds after the called person picked up
the phone

In FS I currently initiate calls like this:

session.waitForAnswer(10000);

if (session.ready()) {
    session.sleep(2000);
    session.streamFile(/*...*/);
}

Is this right?


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From carole.olivier at enst.fr  Fri Dec 12 06:26:27 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Fri, 12 Dec 2008 06:26:27 -0800 (PST)
Subject: [Freeswitch-users] conference_auto_outcall_announce
In-Reply-To: <20956587.post@talk.nabble.com>
References: <20955216.post@talk.nabble.com>
	<0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>
	<20956587.post@talk.nabble.com>
Message-ID: <20976612.post@talk.nabble.com>


Hello,

First, I would like to apologize for a mistake I have made: by adding the
following line in the profile
< param name="enter-sound" value="path/to/file.wav" / >
the enter sound is played.
I am sorry for this. I did not hear it because in the case I have been
analyzing the members of the conference the caller automatically invites are
VoIP speakers which beep before playing anything and apparently miss the
enter sound. (both the beep and the enter-sound have about the same length).

I still have the following questions:
1- Is it possible to introduce a delay so that the enter sound is played
only after 2s?

2- I have noticed that if the caller of the conference talks or makes some
noises at the very beginning when he is entering the conference and the
enter sound is played, we can hear it through the VoIP speakers. Is there
any way to prevent from this? I would like to mute the caller during the
enter-sound and I would need this to be done statically, I mean in the xml
files, and not from the shell.

Thanks!!
Carole



Carole O. wrote:
> 
> Hello,
> 
> Actually, I have already tried it but nothing happens: the file is not
> played and there is no error. 
> There is still a difference: if I configure it as you said, I can not be
> listening anymore, there is simply nothing.
> 
> Would you have an idea? I have checked the path and the syntax 1 million
> times so I do not think I make mistake there.
> 
> Thanks,
> Carole
> 
> 
> 
> Brian West-3 wrote:
>> 
>> Don't have play: in there and it should be fine.  Also if you want the  
>> absolute path you start it with /path/to/file.wav
>> 
>> 
>> /b
>> 
>> On Dec 11, 2008, at 7:13 AM, Carole O. wrote:
>> 
>>> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
>>> [System error : no such file or directory]
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 

-- 
View this message in context: http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20976612.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Fri Dec 12 06:54:42 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 08:54:42 -0600
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
In-Reply-To: <8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
Message-ID: <8329151B-91CE-4F18-99C3-1567DD82D210@freeswitch.org>

FreeSWITCHers,
	I would like to thank everyone that donated.  Enough was raised to  
cover the sound order.  ;)

Thanks,
Brian West
FreeSWITCH.org



From helmut.kuper at ewetel.de  Fri Dec 12 07:01:26 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 12 Dec 2008 16:01:26 +0100
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
Message-ID: <49427CC6.2090407@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
server in a non-root environment.

We experienced a timer problem which led to this FS console error message:

[ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries


During anylizing this we found that q921 T203 is never reset when link
is in state "Multiple Frame Mode Established" and SABME frames are
received by FS. So it must timeout regardless if SABME frames are
received or not.
Additionally we found that the default T203 value (10 sec) was too short
for AVAYA (it has to be >=19 sec)

To fix the problem we changed two things in q921.c:

Change T203 default value from 10 sec to 20000 sec
Line 406: trunk->T203Timeout = 20000;

Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
received SABME frame
Line 1996: Q921T203TimerReset(trunk, tei);

After recompiling FS the Error disapeared. Next week we will do some
calls over the link to make sure there are no other side effects.

Is it planned to make the q921 timeouts configurable in openzap.conf or
in openzap.conf.xml?

best regards
Helmut


PS: My openzap configs:

openzap.conf

[span wanpipe PRI_1]
trunk_type => E1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31




openzap.conf.xml

<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="7"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
   <pri_spans>
     <span name="PRI_1">
       <!-- Log Levels: none, alert, crit, err, warning, notice, info,
debug -->
       <param name="q921loglevel" value="debug"/>
       <param name="q931loglevel" value="debug"/>
       <param name="mode" value="user"/>
       <param name="dialect" value="Oh this is not my dialect"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="default"/>
     </span>
   </pri_spans>
</configuration>

Very interesting here is, that the "dialect" parameter doesn't seem to
have an effect on FS. I use that one above without any errors or warning
and I guess that was not intended.



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iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
=geXp
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From vkobashi at ydeasolutions.com.br  Fri Dec 12 08:15:33 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Fri, 12 Dec 2008 13:15:33 -0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <494198F3.10806@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>
	<494198F3.10806@ydeasolutions.com.br>
Message-ID: <49428E25.80209@ydeasolutions.com.br>

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From anthony.minessale at gmail.com  Fri Dec 12 07:15:25 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2008 09:15:25 -0600
Subject: [Freeswitch-users] fifo.conf.xml usage
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE027E3B3E6E@SBS2008SERVER.consiglia.local>
References: <AclcXcuC05uCObneTKeV2P/QtO/NXQ==>
	<DE9AFF5DF0C8BB4881F86E50EAC0A9AE027E3B3E6E@SBS2008SERVER.consiglia.local>
Message-ID: <191c3a030812120715y5ad0b0e9i639006a32d72afea@mail.gmail.com>

the entries are standard originate strings so all of the {} variables apply.


On Fri, Dec 12, 2008 at 7:30 AM, Jon Bruel <jbr at consiglia.dk> wrote:

>  I'm happy to see that you can add consumers to queues using the
> fifo.conf.xml configuration file. I have made some tests and I hope it may
> lead to a more universal way of setting up queues for small organisations
> than the one I have described in the wiki, and which includes (too) many
> javascripts. I have some questions to clarify my understanding. Using the
> fifo.conf.xml, I find:
>
>  - That the consumers continue to ring after the caller has abandoned the
> queue. Is there a way to avoid this?
>
> Further:
>
>  - Is there a way to control the caller_id_name/number presented to the
> consumer?
>
>  - Is there a way to control the ringing tone in the consumers like the one
> which can be used in the dialplan?
>
>  - Can the fifo.conf.xml refer to an ODBC connection in order to get the
> members from a database?
>
> Finally, thanks for all the good work everybody in the FS community has put
> into FS, I truly believe in the possibilities of this product. Checking the
> hits on Google certainly indicates you moving into the right direction. /Jon
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Dec 12 07:18:09 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2008 09:18:09 -0600
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <49427CC6.2090407@ewetel.de>
References: <49427CC6.2090407@ewetel.de>
Message-ID: <191c3a030812120718n7d8c5410y2ad3cebab8f5be3b@mail.gmail.com>

if you open a jira issue on it we can probably add your patch and/or the
config option.
the users-list is a tough place to manage TDM issues.


On Fri, Dec 12, 2008 at 9:01 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
> AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
> server in a non-root environment.
>
> We experienced a timer problem which led to this FS console error message:
>
> [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries
>
>
> During anylizing this we found that q921 T203 is never reset when link
> is in state "Multiple Frame Mode Established" and SABME frames are
> received by FS. So it must timeout regardless if SABME frames are
> received or not.
> Additionally we found that the default T203 value (10 sec) was too short
> for AVAYA (it has to be >=19 sec)
>
> To fix the problem we changed two things in q921.c:
>
> Change T203 default value from 10 sec to 20000 sec
> Line 406: trunk->T203Timeout = 20000;
>
> Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
> received SABME frame
> Line 1996: Q921T203TimerReset(trunk, tei);
>
> After recompiling FS the Error disapeared. Next week we will do some
> calls over the link to make sure there are no other side effects.
>
> Is it planned to make the q921 timeouts configurable in openzap.conf or
> in openzap.conf.xml?
>
> best regards
> Helmut
>
>
> PS: My openzap configs:
>
> openzap.conf
>
> [span wanpipe PRI_1]
> trunk_type => E1
> b-channel => 1:1-15
> d-channel => 1:16
> b-channel => 1:17-31
>
>
>
>
> openzap.conf.xml
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="7"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>   <pri_spans>
>     <span name="PRI_1">
>       <!-- Log Levels: none, alert, crit, err, warning, notice, info,
> debug -->
>       <param name="q921loglevel" value="debug"/>
>       <param name="q931loglevel" value="debug"/>
>       <param name="mode" value="user"/>
>       <param name="dialect" value="Oh this is not my dialect"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>     </span>
>   </pri_spans>
> </configuration>
>
> Very interesting here is, that the "dialect" parameter doesn't seem to
> have an effect on FS. I use that one above without any errors or warning
> and I guess that was not intended.
>
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
> BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
> =geXp
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Dec 12 07:22:53 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2008 09:22:53 -0600
Subject: [Freeswitch-users] conference_auto_outcall_announce
In-Reply-To: <20976612.post@talk.nabble.com>
References: <20955216.post@talk.nabble.com>
	<0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>
	<20956587.post@talk.nabble.com> <20976612.post@talk.nabble.com>
Message-ID: <191c3a030812120722y7749c160x28e13474ed878943@mail.gmail.com>

No,
there is currently no way.


On Fri, Dec 12, 2008 at 8:26 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> Hello,
>
> First, I would like to apologize for a mistake I have made: by adding the
> following line in the profile
> < param name="enter-sound" value="path/to/file.wav" / >
> the enter sound is played.
> I am sorry for this. I did not hear it because in the case I have been
> analyzing the members of the conference the caller automatically invites
> are
> VoIP speakers which beep before playing anything and apparently miss the
> enter sound. (both the beep and the enter-sound have about the same
> length).
>
> I still have the following questions:
> 1- Is it possible to introduce a delay so that the enter sound is played
> only after 2s?
>
> 2- I have noticed that if the caller of the conference talks or makes some
> noises at the very beginning when he is entering the conference and the
> enter sound is played, we can hear it through the VoIP speakers. Is there
> any way to prevent from this? I would like to mute the caller during the
> enter-sound and I would need this to be done statically, I mean in the xml
> files, and not from the shell.
>
> Thanks!!
> Carole
>
>
>
> Carole O. wrote:
> >
> > Hello,
> >
> > Actually, I have already tried it but nothing happens: the file is not
> > played and there is no error.
> > There is still a difference: if I configure it as you said, I can not be
> > listening anymore, there is simply nothing.
> >
> > Would you have an idea? I have checked the path and the syntax 1 million
> > times so I do not think I make mistake there.
> >
> > Thanks,
> > Carole
> >
> >
> >
> > Brian West-3 wrote:
> >>
> >> Don't have play: in there and it should be fine.  Also if you want the
> >> absolute path you start it with /path/to/file.wav
> >>
> >>
> >> /b
> >>
> >> On Dec 11, 2008, at 7:13 AM, Carole O. wrote:
> >>
> >>> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
> >>> [System error : no such file or directory]
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20976612.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jaugenstine at gmail.com  Fri Dec 12 09:08:41 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Fri, 12 Dec 2008 09:08:41 -0800
Subject: [Freeswitch-users]  call transfer question
Message-ID: <207e7a5e0812120908w1307ee18j5288015132ed3f3e@mail.gmail.com>

I have a call scenario that involves transferring the call and dropping out
of the SIP/RTP stream.  I need to accept the SIP call, play a prompt, and
retrieve a pin code.  After a database lookup, I need to transfer the call
to another FS server and drop out of the SIP path.  I have done this with
the RTP media stream previously.  I am not sure what I need to do to drop
out of the SIP path.  Is this possible on FS?

Jonathan
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From brian at freeswitch.org  Fri Dec 12 09:14:59 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 11:14:59 -0600
Subject: [Freeswitch-users] call transfer question
In-Reply-To: <207e7a5e0812120908w1307ee18j5288015132ed3f3e@mail.gmail.com>
References: <207e7a5e0812120908w1307ee18j5288015132ed3f3e@mail.gmail.com>
Message-ID: <F4FB83DA-79C2-4E68-928A-223E2883B74C@freeswitch.org>

You can use deflect to accomplish this.. it will do a refer to the  
other FS box.

/b

On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote:

> I have a call scenario that involves transferring the call and  
> dropping out of the SIP/RTP stream.  I need to accept the SIP call,  
> play a prompt, and retrieve a pin code.  After a database lookup, I  
> need to transfer the call to another FS server and drop out of the  
> SIP path.  I have done this with the RTP media stream previously.  I  
> am not sure what I need to do to drop out of the SIP path.  Is this  
> possible on FS?
>
> Jonathan



From jaugenstine at gmail.com  Fri Dec 12 09:36:30 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Fri, 12 Dec 2008 09:36:30 -0800
Subject: [Freeswitch-users] call transfer question
In-Reply-To: <F4FB83DA-79C2-4E68-928A-223E2883B74C@freeswitch.org>
References: <207e7a5e0812120908w1307ee18j5288015132ed3f3e@mail.gmail.com>
	<F4FB83DA-79C2-4E68-928A-223E2883B74C@freeswitch.org>
Message-ID: <207e7a5e0812120936m35ac7554mc498d85c003fe282@mail.gmail.com>

Thank you, that is exactly what I need.

On Fri, Dec 12, 2008 at 9:14 AM, Brian West <brian at freeswitch.org> wrote:

> You can use deflect to accomplish this.. it will do a refer to the
> other FS box.
>
> /b
>
> On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote:
>
> > I have a call scenario that involves transferring the call and
> > dropping out of the SIP/RTP stream.  I need to accept the SIP call,
> > play a prompt, and retrieve a pin code.  After a database lookup, I
> > need to transfer the call to another FS server and drop out of the
> > SIP path.  I have done this with the RTP media stream previously.  I
> > am not sure what I need to do to drop out of the SIP path.  Is this
> > possible on FS?
> >
> > Jonathan
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Fri Dec 12 01:38:59 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 12 Dec 2008 20:38:59 +1100
Subject: [Freeswitch-users] Error loading portaudio module
Message-ID: <20081212093859.GA7067@jdc.jasonjgw.net>

I am new to FreeSWITCH; hence this is the first of what will probably be a
number of questions as I learn.

I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
2.6.27, x86_64 architecture), with the portaudio19-dev package installed.

Whenever I try to load the portaudio module I get the following in the logs. I
haven't changed anything in the default portaudio configuration that comes
with FreeSWITCH.

PortAudio version number = 1899
PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
Number of devices = 0
2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
input
 device!
2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
input
 device!
2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
switch_loadable_module_l
oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so

Other software that uses portaudio is known to work. I would expect FreeSWITCH
to detect my Alsa sound devices.

Suggestions welcome.



From alex at sinapticode.ro  Fri Dec 12 10:37:26 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 20:37:26 +0200
Subject: [Freeswitch-users] CDR logs - adding a custom field
Message-ID: <1229107046.4100.18.camel@gathern.lan>

In Asterisk I was able to set a custom CDR field by doing something
like: 
   Set(CDR(userfield)=${SOMETHING})

I need to set a custom field in FreeSwitch, and preferably I want to
have control over its value from Javascript.

Can someone tell me how? :)

Thanks,

-- 
Alexandru Nedelcu
Software Developer, Sinapticode



From anthony.minessale at gmail.com  Fri Dec 12 11:18:32 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2008 13:18:32 -0600
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <1229107046.4100.18.camel@gathern.lan>
References: <1229107046.4100.18.camel@gathern.lan>
Message-ID: <191c3a030812121118y3c339dc1q6697ff8905576c94@mail.gmail.com>

Yes, I'm familiar with that since i invented that feature for Asterisk =D


In FreeSWITCH, All variables are already available from the cdr
just set regular channel variables.

for xml cdr they are all there right away
for csv cdr you can reference any channel variable in your template.




On Fri, Dec 12, 2008 at 12:37 PM, Alexandru Nedelcu <alex at sinapticode.ro>wrote:

> In Asterisk I was able to set a custom CDR field by doing something
> like:
>   Set(CDR(userfield)=${SOMETHING})
>
> I need to set a custom field in FreeSwitch, and preferably I want to
> have control over its value from Javascript.
>
> Can someone tell me how? :)
>
> Thanks,
>
> --
> Alexandru Nedelcu
> Software Developer, Sinapticode
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Dec 12 11:50:10 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 11:50:10 -0800
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <1229107046.4100.18.camel@gathern.lan>
References: <1229107046.4100.18.camel@gathern.lan>
Message-ID: <87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>

Are you using CSV or XML? The reason I ask is because I personally use
XML and I find that having lots of information (even too much) is
better than not enough. The only drawback to XML that I find is that
you have to know how to parse it properly. :) The level of detail in
the XML CDRs is unmatched by any telephony system I've ever
encountered. I highly recommend it.

Also, check out this wiki page if you haven't already:
http://wiki.freeswitch.org/wiki/Mod_xml_cdr

-MC

On Fri, Dec 12, 2008 at 10:37 AM, Alexandru Nedelcu <alex at sinapticode.ro> wrote:
> In Asterisk I was able to set a custom CDR field by doing something
> like:
>   Set(CDR(userfield)=${SOMETHING})
>
> I need to set a custom field in FreeSwitch, and preferably I want to
> have control over its value from Javascript.
>
> Can someone tell me how? :)
>
> Thanks,
>
> --
> Alexandru Nedelcu
> Software Developer, Sinapticode
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Fri Dec 12 11:58:13 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 11:58:13 -0800
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081212093859.GA7067@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>

Jason,

If I understand correctly software other than PA can lock up the sound
card so that PA doesn't "see" it. That might explain why PA reports
number of devices = 0. Could you check to see if possibly something
else has control of your sound card, perhaps ALSA? Turn off anything
that might use the sound card and then restart FS to see if PA can
then detect your device.

-MC

On Fri, Dec 12, 2008 at 1:38 AM, Jason White <jason at jasonjgw.net> wrote:
> I am new to FreeSWITCH; hence this is the first of what will probably be a
> number of questions as I learn.
>
> I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
> 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.
>
> Whenever I try to load the portaudio module I get the following in the logs. I
> haven't changed anything in the default portaudio configuration that comes
> with FreeSWITCH.
>
> PortAudio version number = 1899
> PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
> Number of devices = 0
> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
> input
>  device!
> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
> input
>  device!
> 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
> switch_loadable_module_l
> oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so
>
> Other software that uses portaudio is known to work. I would expect FreeSWITCH
> to detect my Alsa sound devices.
>
> Suggestions welcome.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From alex at sinapticode.ro  Fri Dec 12 12:12:21 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 22:12:21 +0200
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <191c3a030812121118y3c339dc1q6697ff8905576c94@mail.gmail.com>
References: <1229107046.4100.18.camel@gathern.lan>
	<191c3a030812121118y3c339dc1q6697ff8905576c94@mail.gmail.com>
Message-ID: <1229112741.4100.20.camel@gathern.lan>

On Fri, 2008-12-12 at 13:18 -0600, Anthony Minessale wrote:
> Yes, I'm familiar with that since i invented that feature for Asterisk
> =D
> 
> 
> In FreeSWITCH, All variables are already available from the cdr
> just set regular channel variables.
> 
> for xml cdr they are all there right away
> for csv cdr you can reference any channel variable in your template.
> 

Thank you Anthony,

In case someone wants to know how to set channel variables, there's a
link on the wiki here: http://wiki.freeswitch.org/wiki/Channel_Variables




From alex at sinapticode.ro  Fri Dec 12 12:14:33 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 22:14:33 +0200
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
Message-ID: <1229112873.4100.23.camel@gathern.lan>

Thanks Michael,

I'm going to use XML, since I don't really know what variables I want.
Another problem with CSV is that many people parse them with regular
expressions and scripts break when you add a new column.


On Fri, 2008-12-12 at 11:50 -0800, Michael Collins wrote:
> Are you using CSV or XML? The reason I ask is because I personally use
> XML and I find that having lots of information (even too much) is
> better than not enough. The only drawback to XML that I find is that
> you have to know how to parse it properly. :) The level of detail in
> the XML CDRs is unmatched by any telephony system I've ever
> encountered. I highly recommend it.



From alex at sinapticode.ro  Fri Dec 12 12:17:44 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 22:17:44 +0200
Subject: [Freeswitch-users] Freeswitch logging
In-Reply-To: <200812130126.56386.hads@nice.net.nz>
References: <1229083829.4100.11.camel@gathern.lan>
	<200812130126.56386.hads@nice.net.nz>
Message-ID: <1229113064.4100.26.camel@gathern.lan>

On Sat, 2008-12-13 at 01:26 +1300, Hadley Rich wrote:
> This was answered on IRC and a note added to the mod_cdr wiki page.

Thanks Hadley,

I'm a total newbie to FreeSwitch and voip in general, sorry for my
persistence :) I'll try writing an article about my setup this weekend.




From brian at freeswitch.org  Fri Dec 12 12:21:56 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 14:21:56 -0600
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <1229112873.4100.23.camel@gathern.lan>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
Message-ID: <DABF94C5-A8D6-4779-8097-80C16F7F7E6C@freeswitch.org>

What I think would be neat is to have a perl script to parse the XML  
cdr and spit out a graphic of the call path... now that would be neat.
/b

On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote:

> Thanks Michael,
>
> I'm going to use XML, since I don't really know what variables I want.
> Another problem with CSV is that many people parse them with regular
> expressions and scripts break when you add a new column.



From msc at freeswitch.org  Fri Dec 12 12:22:03 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 12:22:03 -0800
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <1229112873.4100.23.camel@gathern.lan>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
Message-ID: <87f2f3b90812121222v7bd1868id093fa6dbe6c368e@mail.gmail.com>

On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu <alex at sinapticode.ro> wrote:
> Thanks Michael,
>
> I'm going to use XML, since I don't really know what variables I want.
> Another problem with CSV is that many people parse them with regular
> expressions and scripts break when you add a new column.
>

This is true. If you build a proper parser for your XML it will easily
be able to handle new channel variables.
-MC


From gmaruzz at celliax.org  Fri Dec 12 12:25:32 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 12 Dec 2008 21:25:32 +0100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
Message-ID: <7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>

Hi there,

you have to use the "default" ALSA audio device to share it, and to
have it automatically format and rate converted.

the "default" ALSA device is not the default portaudio device (not in
the portaudio version used currently by FS).

You have to find out what device id it has under portaudio.

But in this specific case, no device at all was found.

So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA


Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Fri, Dec 12, 2008 at 8:58 PM, Michael Collins <msc at freeswitch.org> wrote:
> Jason,
>
> If I understand correctly software other than PA can lock up the sound
> card so that PA doesn't "see" it. That might explain why PA reports
> number of devices = 0. Could you check to see if possibly something
> else has control of your sound card, perhaps ALSA? Turn off anything
> that might use the sound card and then restart FS to see if PA can
> then detect your device.
>
> -MC
>
> On Fri, Dec 12, 2008 at 1:38 AM, Jason White <jason at jasonjgw.net> wrote:
>> I am new to FreeSWITCH; hence this is the first of what will probably be a
>> number of questions as I learn.
>>
>> I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
>> 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.
>>
>> Whenever I try to load the portaudio module I get the following in the logs. I
>> haven't changed anything in the default portaudio configuration that comes
>> with FreeSWITCH.
>>
>> PortAudio version number = 1899
>> PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
>> Number of devices = 0
>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
>> input
>>  device!
>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
>> input
>>  device!
>> 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
>> switch_loadable_module_l
>> oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so
>>
>> Other software that uses portaudio is known to work. I would expect FreeSWITCH
>> to detect my Alsa sound devices.
>>
>> Suggestions welcome.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sicfslist at gmail.com  Fri Dec 12 12:28:02 2008
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Fri, 12 Dec 2008 14:28:02 -0600
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <87f2f3b90812121222v7bd1868id093fa6dbe6c368e@mail.gmail.com>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
	<87f2f3b90812121222v7bd1868id093fa6dbe6c368e@mail.gmail.com>
Message-ID: <35b355e90812121228k5b10aa57k2928849135f5afdb@mail.gmail.com>

Are there any good examples floating around of XML parsers for this to dump
to MySQL?

On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins <msc at freeswitch.org> wrote:

> On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu <alex at sinapticode.ro>
> wrote:
> > Thanks Michael,
> >
> > I'm going to use XML, since I don't really know what variables I want.
> > Another problem with CSV is that many people parse them with regular
> > expressions and scripts break when you add a new column.
> >
>
> This is true. If you build a proper parser for your XML it will easily
> be able to handle new channel variables.
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alex at sinapticode.ro  Fri Dec 12 12:27:25 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 22:27:25 +0200
Subject: [Freeswitch-users] Configuring FreeSwitch
In-Reply-To: <494129E2.5010602@freeswitch.org>
References: <1228987267.4843.6.camel@gathern.lan>
	<494129E2.5010602@freeswitch.org>
Message-ID: <1229113645.4100.34.camel@gathern.lan>

On Thu, 2008-12-11 at 09:55 -0500, Raymond Chandler wrote:
> 
> i think i answered all of this for you on irc yesterday....
> 

Yes you did, thanks for your help.
I'm a total newbie, but the good news is that I'm almost finished with
my setup. FS is great :)

> use the bridge dialplan app to dial by ip similar to the following:
>     <action application="bridge" 
> data="sofia/${use_profile}/number at ip.address"/>

I'm using "originate" initially. And I think I did something stupid. Is
there anything wrong with the following code ...

var new_session = new Session();
new_session.originate(session, URL);
bridge(session, new_session);

> http://wiki.freeswitch.org/wiki/Sofia#Syntax might also help you out
> a 
> little

It worked great. Thanks.



From gmaruzz at celliax.org  Fri Dec 12 12:30:16 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 12 Dec 2008 21:30:16 +0100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
Message-ID: <7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>

Sorry, the previous one was sent by mistake.

This one is complete:


Hi there,

you have to use the "default" ALSA audio device to share it, and to
have it automatically format and rate converted.

the "default" ALSA device is not the default portaudio device (not in
the portaudio version used currently by FS).

You have to find out what device id it has under portaudio.

But in this specific case, no device at all was found.

So, maybe portaudio was not commpiled with ALSA support (do you have
the ALSA development library installed?).

Also, after recompiling portaudio and mod_portaudio, you can launch FS
giving it the PA_ALSA_PLUGHW=1 environment variable, so portaudio will
use the plughw devices (that are automatically converted to the
desired rate/format) and not the raw devices.

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Fri, Dec 12, 2008 at 9:25 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Hi there,
>
> you have to use the "default" ALSA audio device to share it, and to
> have it automatically format and rate converted.
>
> the "default" ALSA device is not the default portaudio device (not in
> the portaudio version used currently by FS).
>
> You have to find out what device id it has under portaudio.
>
> But in this specific case, no device at all was found.
>
> So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> Company : Celliax
> Website: www.celliax.org
> Address : via Pierlombardo 9, 20135 Milano
> Country/Territory : Italy
> Business Email: gmaruzz at celliax dot org
> Cell : 39-347-2665618
> Fax : 39-02-87390039
>
>
>
>
> On Fri, Dec 12, 2008 at 8:58 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Jason,
>>
>> If I understand correctly software other than PA can lock up the sound
>> card so that PA doesn't "see" it. That might explain why PA reports
>> number of devices = 0. Could you check to see if possibly something
>> else has control of your sound card, perhaps ALSA? Turn off anything
>> that might use the sound card and then restart FS to see if PA can
>> then detect your device.
>>
>> -MC
>>
>> On Fri, Dec 12, 2008 at 1:38 AM, Jason White <jason at jasonjgw.net> wrote:
>>> I am new to FreeSWITCH; hence this is the first of what will probably be a
>>> number of questions as I learn.
>>>
>>> I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
>>> 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.
>>>
>>> Whenever I try to load the portaudio module I get the following in the logs. I
>>> haven't changed anything in the default portaudio configuration that comes
>>> with FreeSWITCH.
>>>
>>> PortAudio version number = 1899
>>> PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
>>> Number of devices = 0
>>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
>>> input
>>>  device!
>>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
>>> input
>>>  device!
>>> 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
>>> switch_loadable_module_l
>>> oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so
>>>
>>> Other software that uses portaudio is known to work. I would expect FreeSWITCH
>>> to detect my Alsa sound devices.
>>>
>>> Suggestions welcome.
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From msc at freeswitch.org  Fri Dec 12 12:29:22 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 12:29:22 -0800
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <DABF94C5-A8D6-4779-8097-80C16F7F7E6C@freeswitch.org>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
	<DABF94C5-A8D6-4779-8097-80C16F7F7E6C@freeswitch.org>
Message-ID: <87f2f3b90812121229o6cbf1fb7x58d9fceed3b5238c@mail.gmail.com>

On Fri, Dec 12, 2008 at 12:21 PM, Brian West <brian at freeswitch.org> wrote:
> What I think would be neat is to have a perl script to parse the XML
> cdr and spit out a graphic of the call path... now that would be neat.
> /b

I think that is a great idea. I was kicking that around as an add-on
feature to a simple CDR database. For example, when browsing the db
for calls, you could click a link that says "view call path" and it
would print a nice purty graph/chart of the call flow. I'll put that
on my rainy-day list...

-MC

>
> On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote:
>
>> Thanks Michael,
>>
>> I'm going to use XML, since I don't really know what variables I want.
>> Another problem with CSV is that many people parse them with regular
>> expressions and scripts break when you add a new column.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Fri Dec 12 12:32:44 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 12:32:44 -0800
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <35b355e90812121228k5b10aa57k2928849135f5afdb@mail.gmail.com>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
	<87f2f3b90812121222v7bd1868id093fa6dbe6c368e@mail.gmail.com>
	<35b355e90812121228k5b10aa57k2928849135f5afdb@mail.gmail.com>
Message-ID: <87f2f3b90812121232l48ee68d5wf48565bbb5ea28b2@mail.gmail.com>

I don't know about "good" examples. I just hacked together a perl
script to extract the very specific elements for my application. If
anyone out there has a sample XML-to-db parser that would be very
welcomed...
-MC

On Fri, Dec 12, 2008 at 12:28 PM, Shelby Ramsey <sicfslist at gmail.com> wrote:
> Are there any good examples floating around of XML parsers for this to dump
> to MySQL?
>
> On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu <alex at sinapticode.ro>
>> wrote:
>> > Thanks Michael,
>> >
>> > I'm going to use XML, since I don't really know what variables I want.
>> > Another problem with CSV is that many people parse them with regular
>> > expressions and scripts break when you add a new column.
>> >
>>
>> This is true. If you build a proper parser for your XML it will easily
>> be able to handle new channel variables.
>> -MC
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From pedro2263 at gmail.com  Fri Dec 12 13:47:09 2008
From: pedro2263 at gmail.com (Pedro .)
Date: Fri, 12 Dec 2008 15:47:09 -0600
Subject: [Freeswitch-users] Cepstral SDK
Message-ID: <28b3653a0812121347i740fae06w73065605bc7b6eba@mail.gmail.com>

Hi,

I'm trying to integrate Cepstral TTS I read in the wiki that I need
Ceptral's SDK to compile the mod_ceptral, can somebody tell me where can I
get the trial version of this SDK?.

Thanks.
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From brian at freeswitch.org  Fri Dec 12 14:14:37 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 16:14:37 -0600
Subject: [Freeswitch-users] Cepstral SDK
In-Reply-To: <28b3653a0812121347i740fae06w73065605bc7b6eba@mail.gmail.com>
References: <28b3653a0812121347i740fae06w73065605bc7b6eba@mail.gmail.com>
Message-ID: <726FFD6A-7E1D-4BE9-A96C-2D885BDF5931@freeswitch.org>

If you're on linux you need to go download and install any voice.  If  
you're on windows I have to forward your request to Cepstral to get  
the SDK for windows.

/b

On Dec 12, 2008, at 3:47 PM, Pedro . wrote:

> Hi,
>
> I'm trying to integrate Cepstral TTS I read in the wiki that I need  
> Ceptral's SDK to compile the mod_ceptral, can somebody tell me where  
> can I get the trial version of this SDK?.
>
> Thanks.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason at jasonjgw.net  Fri Dec 12 15:24:03 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 10:24:03 +1100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
Message-ID: <20081212232403.GA7667@jdc.jasonjgw.net>

On Fri, Dec 12, 2008 at 09:30:16PM +0100, Giovanni Maruzzelli wrote:
 
> But in this specific case, no device at all was found.
> 
> So, maybe portaudio was not commpiled with ALSA support (do you have
> the ALSA development library installed?).

Yes, and in any case the version of PortAudio which is installed came from the
Debian package.

Does FreeSWITCH support PortAudio 19? If not, maybe there are API differences.
> 
> Also, after recompiling portaudio and mod_portaudio, you can launch FS
> giving it the PA_ALSA_PLUGHW=1 environment variable, so portaudio will
> use the plughw devices (that are automatically converted to the
> desired rate/format) and not the raw devices.

I'll try that.

To answer another question that arose in this thread, I have no other software
currently using the audio devices. Alsa is known to work, as is other software
that accesses the Alsa devices with PortAudio.



From frank at impactfax.com  Fri Dec 12 15:51:35 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 12 Dec 2008 18:51:35 -0500
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
Message-ID: <026101c95cb4$91ad1d40$33014c0a@ws4>

Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call?
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From brian at freeswitch.org  Fri Dec 12 15:54:44 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 17:54:44 -0600
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <026101c95cb4$91ad1d40$33014c0a@ws4>
References: <026101c95cb4$91ad1d40$33014c0a@ws4>
Message-ID: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>

sched_api  (hint uuid_send_dtmf)

API CALL [sched_api()] output:
-ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>


/b

On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:

> Is there a way to schedule a certain DTMF tone to be played into a  
> bridge (both a and b legs) after a scheduled number of seconds into  
> the call?
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Fri Dec 12 16:50:04 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 11:50:04 +1100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081212232403.GA7667@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
Message-ID: <20081213005004.GA8393@jdc.jasonjgw.net>

This is interesting...

I wrote the following test.


#include<stdio.h>
#include <portaudio.h>
int main(int argc, char **argv) 
{
  Pa_Initialize();
  printf("Number of devices: %d\n",Pa_GetDeviceCount());
  Pa_Terminate();
}

then I compiled and executed it:
gcc -o pa_test pa_test.c -lportaudio
./portaudio

Number of devices: 10

>From what I can see, the code in mod_portaudio.c is using exactly the same API
call, but it seems to be returning 0 in that case.



From jason at jasonjgw.net  Fri Dec 12 17:35:38 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 12:35:38 +1100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081213005004.GA8393@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
Message-ID: <20081213013538.GA16338@jdc.jasonjgw.net>

With apologies for the noise on the list, I just realized that FreeSWITCH is
building its own version of PortAudio.

I can confirm that Alsa is being detected and support for it included.

So, there must be some difference between the version of PortAudio that comes
with FreeSWITCH, and the version installed in my /usr/lib, such that the
FreeSWITCH version fails to detect my Alsa devices.



From ack at telefonica.net  Fri Dec 12 17:58:01 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Sat, 13 Dec 2008 02:58:01 +0100
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <1228875010.2477.67.camel@develop4>
References: <1228352588.25709.42.camel@develop4>
	<191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
	<1228365981.25709.60.camel@develop4>
	<191c3a030812040734s4f514f42s9a30a48c93709fd5@mail.gmail.com>
	<1228875010.2477.67.camel@develop4>
Message-ID: <1229133482.10071.59.camel@develop4>

Thanks again Anthony ! 

You fixed the issue with DTMF i had reported :

http://jira.freeswitch.org/browse/FSCORE-251



Chris Danielson added to Wiki a nice page collecting these issues with
Sonus :

http://wiki.freeswitch.org/wiki/RTP_Issues


Cheers,

El mi?, 10-12-2008 a las 03:10 +0100, Angel Carpintero escribi?:
> Thanks Anthony , you did a great work ! this is fixed in svn r10691.
> 
> Some notes for people using Sonus and L3 as was my case :
> 
> in var.xml in some scenario you may need :
> 
> <X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/>
> 
> in sip_profiles/internal.xml :
> 
> <param name="rtp-rewrite-timestamps" value="true"/>
> 
> might help for some people with rtp issues :
> 
> <param name="rtp-timer-name" value="none"/>
> 
> If you have issues with DTMF and timestamps add also :
> 
> <param name="pass-rfc2833" value="true"/>
> 
> I've a little issues with DTMF from VOIP , i i'll figure out can could
> be the issue , from PSTN all works like a charm :)
> 
> Cheers,
> 
> El jue, 04-12-2008 a las 09:34 -0600, Anthony Minessale escribi?:
> > most likely it's because during the time you are dong artificial
> > ringback the other side is not doing RTP right.
> > 
> > When the call is answered we flush the rtp buffer and your missing
> > audio is probably flushed with it.
> > so you can choose to have a 3 second delay or erase the 3 seconds as
> > it does now.
> > 
> > This is a known problem with sonus which has been proven to build up
> > an audio delay during the time
> > you are waiting for the call to answer.  I'm sure you prefer the way
> > it is to a large audio delay.
> > 
> > 
> > 
> > On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <ack at telefonica.net>
> > wrote:
> >         No TDM , all is SIP :
> >         
> >         
> >         PSTN ---> Sip Proxy_A --> FS ( brigde )
> >         ringback/transfer_ringback
> >         -> Sip Proxy_B --> PSTN
> >         
> >         
> >         In logfile i think you can get some details about Media
> >         Gateways
> >         ( Sonus ) PSTN inbound / outbound is provided by Level3.
> >         
> >         I can get a capture of a call if you want, in capture the
> >         audio is not
> >         missing, issue with :
> >         
> >         - rtp buffer ?
> >         - Sonus ?
> >         
> >         Let me know anything you need so i can provide a log or create
> >         a new
> >         scenario.
> >         
> >         
> >         Thanks,
> >         
> >         El mi?, 03-12-2008 a las 22:12 -0600, Anthony Minessale
> >         escribi?:
> >         
> >         > what does PSTN represent?
> >         >
> >         > I know what the PSTN is but how are you reaching it?
> >         > is it TDM, SIP etc... what gateway type other details.
> >         >
> >         >
> >         > On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero
> >         <ack at telefonica.net>
> >         > wrote:
> >         >         Hi guys,
> >         >
> >         >          I've a strange issue with FS , version svn
> >         -r10584 ,
> >         >         when FS bridges a call first 3 seconds of audio are
> >         missing ,
> >         >         looks that
> >         >         only happens on PSTN calls and using ringback or
> >         >         transfer_ringback. This
> >         >         only happens in calls from PSTN , not from VOIP.
> >         Some
> >         >         scenarios i tried
> >         >         to isolate this issue :
> >         >
> >         >
> >         >         - Issue
> >         >
> >         >         PSTN --> FS ( brigde ) ringback/transfer_ringback ->
> >         PSTN
> >         >
> >         >         - Good setting bypass_media before run bridge but i
> >         need rtp
> >         >         in FS path
> >         >
> >         >         PSTN --> FS ( brigde ) ringback/transfer_ringback ->
> >         PSTN
> >         >
> >         >         - Good
> >         >
> >         >         PSTN --> FS ( brigde ) WITHOUT
> >         ringback/transfer_ringback ->
> >         >         PSTN
> >         >
> >         >         - Always good
> >         >
> >         >         VOIP --> FS ( brigde ) -> PSTN
> >         >
> >         >
> >         >         Dialplan has nothing wrong ( i guess ):
> >         >
> >         >         <extension name="Transfers">
> >         >            <condition field="destination_number"
> >         >         expression="^1??XXXXXXXXXX$">
> >         >              <action application="answer"/>
> >         >              <action application="speak" data="cepstral|
> >         Allison-8kHz|
> >         >         blah"/>
> >         >              <action application="set"
> >         >         data="hangup_after_bridge=false"/>
> >         >              <action application="set"
> >         data="playback_terminators=#"/>
> >         >              <action application="set" data="ringback=
> >         $${us-ring}"/>
> >         >              <action application="set"
> >         data="transfer_ringback=
> >         >         $${hold_music}"/>
> >         >              <action application="set"
> >         data="effective_caller_id_name=
> >         >         ${caller_id_name}"/>
> >         >              <action application="set"
> >         >         data="effective_caller_id_number=
> >         >         ${caller_id_number}"/>
> >         >              <action application="set"
> >         data="originate_timeout=30"/>
> >         >              <action application="set"
> >         data="call_timeout=30"/>
> >         >              <action application="bridge"
> >         >         data="sofia/default/18008226235 at PSTN_GW"/>
> >         >              <action application="speak" data="cepstral|
> >         Allison-8kHz|
> >         >         Transfer
> >         >         finished"/>
> >         >              <action application="hangup"/>
> >         >            </condition>
> >         >          </extension>
> >         >
> >         >
> >         >
> >         >         Any ideas ?
> >         >
> >         >         Attached log of FS ( F8 from console ).
> >         >
> >         >
> >         >         Thanks in advance !
> >         >
> >         >         --
> >         >         Angel Carpintero
> >         >         ack ( at ) telefonica ( dot ) net
> >         >
> >         >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF
> >         AC2C CA61
> >         >         6EF1 B90D
> >         >
> >         >
> >         >
> >         
> >         > --
> >         > Anthony Minessale II
> >         >
> >         > FreeSWITCH http://www.freeswitch.org/
> >         > ClueCon http://www.cluecon.com/
> >         >
> >         > AIM: anthm
> >         > MSN:anthony_minessale at hotmail.com
> >         > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >         > IRC: irc.freenode.net #freeswitch
> >         >
> >         > FreeSWITCH Developer Conference
> >         > sip:888 at conference.freeswitch.org
> >         > iax:guest at conference.freeswitch.org/888
> >         > googletalk:conf+888 at conference.freeswitch.org
> >         > pstn:213-799-1400
> >         
> >         
> >         --
> >         
> >         Angel Carpintero
> >         ack ( at ) telefonica ( dot ) net
> >         
> >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
> >         6EF1 B90D
> >         
> >         
> >         
> >         _______________________________________________
> >         Freeswitch-users mailing list
> >         Freeswitch-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >         
> > 
> > 
> > 
> > -- 
> > Anthony Minessale II
> > 
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > 
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> > 
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:213-799-1400
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

"No basta saber, hay que aplicar lo que se sabe; 
no basta querer hacerlas cosas, hay que hacerlas".

"Knowing is not enough; we must apply. 
 Willing is not enough; we must do"

 Johann Wolfgang von Goethe
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From jason at jasonjgw.net  Fri Dec 12 18:01:36 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 13:01:36 +1100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081213013538.GA16338@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
	<20081213013538.GA16338@jdc.jasonjgw.net>
Message-ID: <20081213020136.GA16597@jdc.jasonjgw.net>

The problem is now solved.

It turned out to be permissions: the freeswitch user wasn't added to the audio
group in /etc/group, hence didn't have permission to interrogate the audio
devices.

Perhaps a future version of the Debian package could address this, or at least
it should be noted somewhere.



From msc at freeswitch.org  Fri Dec 12 18:25:58 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 18:25:58 -0800
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081213020136.GA16597@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
	<20081213013538.GA16338@jdc.jasonjgw.net>
	<20081213020136.GA16597@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812121825t531a60b6k60a3e57796338578@mail.gmail.com>

Jason,

Thanks for troubleshooting this! At the very least I will add a note
to the PA section of the wiki.
-MC

On Fri, Dec 12, 2008 at 6:01 PM, Jason White <jason at jasonjgw.net> wrote:
> The problem is now solved.
>
> It turned out to be permissions: the freeswitch user wasn't added to the audio
> group in /etc/group, hence didn't have permission to interrogate the audio
> devices.
>
> Perhaps a future version of the Debian package could address this, or at least
> it should be noted somewhere.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From frank at impactfax.com  Fri Dec 12 19:37:20 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 12 Dec 2008 22:37:20 -0500
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>
Message-ID: <02b601c95cd4$1b093460$33014c0a@ws4>

Not much written in the wiki on this.  Also searched the list and not
much on either sched_api or uuid_send_dtmf.
So from an xml dialplan, can sched_api as an application?
Is there any way to have the time offset reference the point at which
the call started ?  ie. When the called party answers?
 
Ultimately, I was trying to insert some xml into my dial plan that would
play a dtmf tone 10 seconds after the called party picked up the phone.
But from the little that has been written so far that I can find, it is
not clear to me how to piece this together.  Am I being dense and
missing anything that has already been written?
 
/f
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West


sched_api  (hint uuid_send_dtmf)
 
API CALL [sched_api()] output:
-ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>
 
 
/b
 
On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:



Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call?
_______________________________________________
 
 
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From msc at freeswitch.org  Fri Dec 12 20:21:27 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 20:21:27 -0800
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <02b601c95cd4$1b093460$33014c0a@ws4>
References: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>
	<02b601c95cd4$1b093460$33014c0a@ws4>
Message-ID: <87f2f3b90812122021y2979d21anadab85a70ed4717e@mail.gmail.com>

Frank,

I'm sure this is possible. Please give me a little bit to look into
this. I'm going to see if I can lab it up and give you a sample
dialplan. Also, thanks for the heads up on the wiki not having this
information. I will put that on my not-so-short wiki todo list.

Thanks,
MC

On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact <frank at impactfax.com> wrote:
> Not much written in the wiki on this.  Also searched the list and not much
> on either sched_api or uuid_send_dtmf.
>
> So from an xml dialplan, can sched_api as an application?
>
> Is there any way to have the time offset reference the point at which the
> call started ?  ie. When the called party answers?
>
>
>
> Ultimately, I was trying to insert some xml into my dial plan that would
> play a dtmf tone 10 seconds after the called party picked up the phone.  But
> from the little that has been written so far that I can find, it is not
> clear to me how to piece this together.  Am I being dense and missing
> anything that has already been written?
>
>
>
> /f
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
>
> sched_api  (hint uuid_send_dtmf)
>
>
>
> API CALL [sched_api()] output:
>
> -ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>
>
>
>
>
>
> /b
>
>
>
> On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:
>
> Is there a way to schedule a certain DTMF tone to be played into a bridge
> (both a and b legs) after a scheduled number of seconds into the call?
>
> _______________________________________________
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Dec 12 22:39:21 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 22:39:21 -0800
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <02b601c95cd4$1b093460$33014c0a@ws4>
References: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>
	<02b601c95cd4$1b093460$33014c0a@ws4>
Message-ID: <87f2f3b90812122239y127f35d6r80cd1286b7dddae8@mail.gmail.com>

Frank,

I found a simple way to handle this scenario. I decided just to create
a small Lua script that would do the job. It's committed in latest
trunk. Look in src/scripts/contrib/mcollins for uuid_send_dtmf.lua. It
has comments on how to call it, including a sample dp call.

The way I would use this in your scenario is to setup a destination
using the execute_on_answer variable.
http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer

Have the destination be an extension that does something like this:
<extension...>
  <condition...>
    <action application="answer"/>
    <action application="lua" data="uuid_send_dtmf ${uuid} 10 123"/>
    ...rest of diaplan...
  </condition>
</extension>

The 10 means ten seconds, the 123 means send the dtmf digits 1,2,3 in
order. You can tinker with the settings as you see fit.

Please let me know how it goes. BTW, be sure to put the Lua script in
/usr/local/freeswitch/scripts or specify the complete path name when
calling the lua app in the dialplan.

-MC

On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact <frank at impactfax.com> wrote:
> Not much written in the wiki on this.  Also searched the list and not much
> on either sched_api or uuid_send_dtmf.
>
> So from an xml dialplan, can sched_api as an application?
>
> Is there any way to have the time offset reference the point at which the
> call started ?  ie. When the called party answers?
>
>
>
> Ultimately, I was trying to insert some xml into my dial plan that would
> play a dtmf tone 10 seconds after the called party picked up the phone.  But
> from the little that has been written so far that I can find, it is not
> clear to me how to piece this together.  Am I being dense and missing
> anything that has already been written?
>
>
>
> /f
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
>
> sched_api  (hint uuid_send_dtmf)
>
>
>
> API CALL [sched_api()] output:
>
> -ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>
>
>
>
>
>
> /b
>
>
>
> On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:
>
> Is there a way to schedule a certain DTMF tone to be played into a bridge
> (both a and b legs) after a scheduled number of seconds into the call?
>
> _______________________________________________
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Dec 12 22:41:00 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 22:41:00 -0800
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <87f2f3b90812122021y2979d21anadab85a70ed4717e@mail.gmail.com>
References: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>
	<02b601c95cd4$1b093460$33014c0a@ws4>
	<87f2f3b90812122021y2979d21anadab85a70ed4717e@mail.gmail.com>
Message-ID: <87f2f3b90812122241t5dfa8bdeif15299684666c12e@mail.gmail.com>

On Fri, Dec 12, 2008 at 8:21 PM, Michael Collins <msc at freeswitch.org> wrote:
> Frank,
>
> I'm sure this is possible. Please give me a little bit to look into
> this. I'm going to see if I can lab it up and give you a sample
> dialplan. Also, thanks for the heads up on the wiki not having this
> information. I will put that on my not-so-short wiki todo list.
>
> Thanks,
> MC
>
> On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact <frank at impactfax.com> wrote:
>> Not much written in the wiki on this.  Also searched the list and not much
>> on either sched_api or uuid_send_dtmf.
>>
>> So from an xml dialplan, can sched_api as an application?
>>
>> Is there any way to have the time offset reference the point at which the
>> call started ?  ie. When the called party answers?
>>
>>
>>
>> Ultimately, I was trying to insert some xml into my dial plan that would
>> play a dtmf tone 10 seconds after the called party picked up the phone.  But
>> from the little that has been written so far that I can find, it is not
>> clear to me how to piece this together.  Am I being dense and missing
>> anything that has already been written?
>>
>>
>>
>> /f
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
>> West
>>
>> sched_api  (hint uuid_send_dtmf)
>>
>>
>>
>> API CALL [sched_api()] output:
>>
>> -ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>
>>
>>
>>
>>
>>
>> /b
>>
>>
>>
>> On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:
>>
>> Is there a way to schedule a certain DTMF tone to be played into a bridge
>> (both a and b legs) after a scheduled number of seconds into the call?
>>
>> _______________________________________________
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From mszlazak at aol.com  Sat Dec 13 00:05:55 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Sat, 13 Dec 2008 03:05:55 -0500
Subject: [Freeswitch-users] Confirmation checking on VAD is needed.
Message-ID: <8CB2B10C221A1AB-4C4-1AB5@WEBMAIL-MY18.sysops.aol.com>

I need confirmation on whether VAD is disabled in the FS default configuration. I've looked at conf\sip_profiles\internal.xml and the relevant section was the following which seems to indicate that VAD is off by default since I don't see VAD value="none".

??? <!-- VAD choose one (out is a good choice); -->
??? <!-- <param name="vad" value="in"/> -->
??? <!-- <param name="vad" value="out"/> -->
??? <!-- <param name="vad" value="both"/> -->
??? <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
??? <!--all inbound reg will look in this domain for the users -->
??? <!--<param name="force-register-domain" value="cluecon.com"/>-->
??? <!-- disable register and transfer which may be undesirable in a public switch -->
??? <!--<param name="disable-transfer" value="true"/>-->
??? <!--<param name="disable-register" value="true"/>-->

However, is there a way to confirm this in the FS console?
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From jason at jasonjgw.net  Sat Dec 13 00:46:43 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 19:46:43 +1100
Subject: [Freeswitch-users] Confirmation checking on VAD is needed.
In-Reply-To: <8CB2B10C221A1AB-4C4-1AB5@WEBMAIL-MY18.sysops.aol.com>
References: <8CB2B10C221A1AB-4C4-1AB5@WEBMAIL-MY18.sysops.aol.com>
Message-ID: <20081213084643.GA2378@jdc.jasonjgw.net>

On Sat, Dec 13, 2008 at 03:05:55AM -0500, mszlazak at aol.com wrote:
> I need confirmation on whether VAD is disabled in the FS default configuration. I've looked at conf\sip_profiles\internal.xml and the relevant section was the following which seems to indicate that VAD is off by default since I don't see VAD value="none".
> 

This is all commented out, and from a quick look at the code in the sofia SIP
module, VAD won't be enabled unless you explicitly set it to "in" or "out".



From faisalmaqsoodi at yahoo.com  Sat Dec 13 01:05:22 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sat, 13 Dec 2008 01:05:22 -0800 (PST)
Subject: [Freeswitch-users] Maintaining call detail record
Message-ID: <831510.71942.qm@web30702.mail.mud.yahoo.com>

How can i interface fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked on xml cdr but it contains too much info, more than i need and in a format which is not easily understandable. I also tried perl coding mentioned on the link at the bottom of the page
http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r displayed during its execution. Is there any easy method for that. Plz help me. 
?????????????????????????????????????????????????????????????????????????????????????? faisal



      
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From faisalmaqsoodi at yahoo.com  Sat Dec 13 01:44:38 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sat, 13 Dec 2008 01:44:38 -0800 (PST)
Subject: [Freeswitch-users] Maintaining call detail record
Message-ID: <614188.90296.qm@web30706.mail.mud.yahoo.com>

                        
                        

if (typeof YAHOO == "undefined") {
 var YAHOO = {};
}
YAHOO.Shortcuts = YAHOO.Shortcuts || {};
YAHOO.Shortcuts.hasSensitiveText = false;
YAHOO.Shortcuts.sensitivityType = [];
YAHOO.Shortcuts.doUlt = false;
YAHOO.Shortcuts.location = "us";
YAHOO.Shortcuts.document_id = 0;
YAHOO.Shortcuts.document_type = "";
YAHOO.Shortcuts.document_title = "Maintaining call detail record";
YAHOO.Shortcuts.document_publish_date = "";
YAHOO.Shortcuts.document_author = "faisalmaqsoodi at yahoo.com";
YAHOO.Shortcuts.document_url = "";
YAHOO.Shortcuts.document_tags = "";
YAHOO.Shortcuts.document_language = "english";
YAHOO.Shortcuts.annotationSet = {
"lw_1229161419_0": {
"text": "caller id",
"extended": 0,
"startchar": 187,
"endchar": 195,
"start": 187,
"end": 195,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 0.2054,
"relScore": 5.33095,
"type": ["shortcuts:/concept"],
"category": ["CONCEPT"],
"wikiId": "Caller_ID",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked",
"metaData": {
"visible": "true"
}
 },
"lw_1229161419_1": {
"text": "http://wiki.freeswitch.org/wiki/Mod_cdr_csv,",
"extended": 0,
"startchar": 425,
"endchar": 468,
"start": 425,
"end": 468,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 1,
"relScore": 0,
"type": ["shortcuts:/us/instance/identifier/URL"],
"category": ["IDENTIFIER"],
"wikiId": "",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "",
"metaData": {
"visible": "true"
}
 }
};
YAHOO.Shortcuts.headerID = "c0b92bae50300eda0a2889f03525f330";

How can i interface fs with mysql in order to maintain calls record like caller id
and time n date of call etc. I ve worked on xml cdr but it contains too
much info, more than i need and in a format which is not easily
understandable. I also tried perl coding mentioned on the link at the
bottom of the page
http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r displayed during its execution. Is there any easy method for that. Plz help me.
 
?????????????????????????????????????????????????????????????????????????????????????? faisal


      
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From dalechase at yahoo.com  Sat Dec 13 11:03:34 2008
From: dalechase at yahoo.com (dalechase at yahoo.com)
Date: Sat, 13 Dec 2008 11:03:34 -0800 (PST)
Subject: [Freeswitch-users] config help: openzap and T1 A102u
References: <603978.74977.qm@web36108.mail.mud.yahoo.com>
	<004001c95c38$04da0210$6d02a8c0@opos20>
Message-ID: <255403.64286.qm@web36104.mail.mud.yahoo.com>

Thanks, that got things going...



----- Original Message ----
From: Evgeniy Zolotov <zolotov at altron.ua>
To: freeswitch-users at lists.freeswitch.org
Sent: Friday, December 12, 2008 1:00:01 AM
Subject: Re: [Freeswitch-users] config help: openzap and T1 A102u

  Did you try ./wanrouter start before starting FreeSWITCH ?


      


From chavpaskov at shaw.ca  Sat Dec 13 12:24:25 2008
From: chavpaskov at shaw.ca (Chav Paskov)
Date: Sat, 13 Dec 2008 12:24:25 -0800
Subject: [Freeswitch-users] how to handle  returned sip 302 dialplan
Message-ID: <494419F9.6090304@shaw.ca>

Good morning everybody,

I would like to know what would be the best  way to process the info in 
returned back sip 302  message.
Let me be a bit more specific.
i'm sending INVITE as pasted below to  z.z.z.z

*INVITE* sip:17402740539 at z.z.z.z:5000 SIP/2.0^M
Via: SIP/2.0/UDP z.z.z.z:5000;branch=z9hG4bK215b.e6ae7e74.0^M
Via: SIP/2.0/UDP 
y.y.y.y:5060;branch=z9hG4bKaa6257694041475e0659eeb0019a0bbc^M
*From: "Bob Caller" <sip:6144102508 at y.y.y.y:5060>;tag=CCC7.1F37^M
To: <sip:17402740539 at x.x.x.x:5060>^M
*Call-ID: 15821337-3367080415-112396 at msc-01.voice.example.net^M
CSeq: 1 INVITE^M
Contact: <sip:6144102508 at y.y.y.y:5060>^M
Content-Type: application/sdp^M
Max-forwards: 69^M
Session-expires: 3600;Refresher=uac^M
Supported: timer^M
Content-Length:   200^M
^M
1192 v=0^M
o=iserver 16123 16124 IN IP4 y.y.y.y^M
s=sip call^M
c=IN IP4 64.112.188.81^M
t=0 0^M
m=audio 55824 RTP/AVP 18 0 101^M
a=sendrecv^M
a=fmtp:101 0-15^M
a=rtpmap:101 telephone-event/8000^M
a=ptime:20^M

the reply from  z.z.z.z back is  sip 302

SIP/2.0 *302 Moved Temporarily*^M
Via: SIP/2.0/UDP z.z.z.z:5000;branch=z9hG4bK215b.e6ae7e74.0^M
Via: SIP/2.0/UDP 
y.y.y.y:5060;branch=z9hG4bKaa6257694041475e0659eeb0019a0bbc^M
From: "Bob Caller" <sip:6144102508 at y.y.y.y:5060>;tag=CCC7.1F37^M
To: <sip:17402740539 at x.x.x.x:5060>;tag=1087415105^M
Call-ID: 15821337-3367080415-112396 at msc-01.voice.example.net^M
CSeq: 1 INVITE^M
*Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
*User-Agent: eXosip/3.1.0^M
Content-Length:  


my question is:
 
Is it possible to send the call to z.z.z.z , receive the SIP 302 , 
process the data in Contact field  and redirect to the new destination 
contained in *Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
*without closing the session.
i red something about   <action application="set" 
data="continue_on_fail=true"/>  but i'm not sure  how to use it.

Any ideas on this matter  will be highly appreciated.
Best Regards
Chav






From msc at freeswitch.org  Sat Dec 13 12:51:12 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 12:51:12 -0800
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <49427CC6.2090407@ewetel.de>
References: <49427CC6.2090407@ewetel.de>
Message-ID: <87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>

On Fri, Dec 12, 2008 at 7:01 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
> AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
> server in a non-root environment.
>
> We experienced a timer problem which led to this FS console error message:
>
> [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries
>
>
> During anylizing this we found that q921 T203 is never reset when link
> is in state "Multiple Frame Mode Established" and SABME frames are
> received by FS. So it must timeout regardless if SABME frames are
> received or not.
> Additionally we found that the default T203 value (10 sec) was too short
> for AVAYA (it has to be >=19 sec)
>
> To fix the problem we changed two things in q921.c:
>
> Change T203 default value from 10 sec to 20000 sec
> Line 406: trunk->T203Timeout = 20000;
>
> Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
> received SABME frame
> Line 1996: Q921T203TimerReset(trunk, tei);
>
> After recompiling FS the Error disapeared. Next week we will do some
> calls over the link to make sure there are no other side effects.
>
> Is it planned to make the q921 timeouts configurable in openzap.conf or
> in openzap.conf.xml?
>
> best regards
> Helmut
>
>
> PS: My openzap configs:
>
> openzap.conf
>
> [span wanpipe PRI_1]
> trunk_type => E1
> b-channel => 1:1-15
> d-channel => 1:16
> b-channel => 1:17-31
>
>
>
>
> openzap.conf.xml
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="7"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>   <pri_spans>
>     <span name="PRI_1">
>       <!-- Log Levels: none, alert, crit, err, warning, notice, info,
> debug -->
>       <param name="q921loglevel" value="debug"/>
>       <param name="q931loglevel" value="debug"/>
>       <param name="mode" value="user"/>
>       <param name="dialect" value="Oh this is not my dialect"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>     </span>
>   </pri_spans>
> </configuration>
>
> Very interesting here is, that the "dialect" parameter doesn't seem to
> have an effect on FS. I use that one above without any errors or warning
> and I guess that was not intended.

At this point in the OZ development we've got it set to default to
"national" if the dialect isn't otherwise properly specified. It does
make sense to throw an error if the dialect is not properly specified,
even if we still default to national.

-MC

>
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
> BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
> =geXp
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From astmac at stillnewt.org  Sat Dec 13 13:00:19 2008
From: astmac at stillnewt.org (martin joseph)
Date: Sat, 13 Dec 2008 13:00:19 -0800
Subject: [Freeswitch-users] (no subject)
Message-ID: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>

I am wondering if anyone running OSX 10.4.11 is able to successfully  
build FS Trunk?

If so,  what versions does build.sh report? Please?

I get the following and have issues making it...

Updated to revision 10753.
mail:/usr/src/freeswitch/trunk root# ./bootstrap.sh
bootstrap: checking installation...
bootstrap: autoconf version 2.61 (ok)
bootstrap: automake version 1.10 (ok)
bootstrap: libtool version 1.5.24 (ok)

Thanks for the info,
Marty



From frank at impactfax.com  Sat Dec 13 13:20:16 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 13 Dec 2008 16:20:16 -0500
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <87f2f3b90812122239y127f35d6r80cd1286b7dddae8@mail.gmail.com>
Message-ID: <03c801c95d68$9880ce50$33014c0a@ws4>

Michael,

Got it working.  Just a little simpler then you outlined.
I just added to my xml dialplan this line.

<action application="set" data="execute_on_answer=lua uuid_send_dtmf.lua
${uuid} 20 123"/>

I added this just before the bridge application.

I did this instead of adding an extra extension to transfer to on
answer.  Everything worked well.  The DTMF was played to the calling
party.  Out of curiosity, if we wanted also to play the DTMF to the
called party also, what would we have to give uuid_send_dtmf?
Particularly since we call it before the bridge.

Can uuid_send_dtmf accept anything like "w" for wait or anything else
special for DTMF stuff?

Also, I got an error output to the console when the sched_api was run.
See below.
******
2008-12-13 16:07:28 [NOTICE] switch_cpp.cpp:1050 console_log() apicmd:
sched_api
2008-12-13 16:07:28 [NOTICE] switch_cpp.cpp:1050 console_log() apiarg:
+20 none uuid_send_dtmf 37618e54-c959-11dd-bc73-0923daa880b2 123

2008-12-13 16:07:28 [ERR] switch_cpp.cpp:1050 console_log() Result is
+OK Added: 49751
******
is this ERR anything to worry about even though we got a result ok?

Thanks again for the help.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
 
Frank,

I found a simple way to handle this scenario. I decided just to create
a small Lua script that would do the job. It's committed in latest
trunk. Look in src/scripts/contrib/mcollins for uuid_send_dtmf.lua. It
has comments on how to call it, including a sample dp call.

The way I would use this in your scenario is to setup a destination
using the execute_on_answer variable.
http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer

Have the destination be an extension that does something like this:
<extension...>
  <condition...>
    <action application="answer"/>
    <action application="lua" data="uuid_send_dtmf ${uuid} 10 123"/>
    ...rest of diaplan...
  </condition>
</extension>

The 10 means ten seconds, the 123 means send the dtmf digits 1,2,3 in
order. You can tinker with the settings as you see fit.

Please let me know how it goes. BTW, be sure to put the Lua script in
/usr/local/freeswitch/scripts or specify the complete path name when
calling the lua app in the dialplan.

-MC





From brian at freeswitch.org  Sat Dec 13 13:28:58 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Dec 2008 15:28:58 -0600
Subject: [Freeswitch-users] how to handle  returned sip 302 dialplan
In-Reply-To: <494419F9.6090304@shaw.ca>
References: <494419F9.6090304@shaw.ca>
Message-ID: <9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>

Chav,
	Once the 302 is received by FreeSWITCH it will follow it to the  
contact listed in the 302.  What else are you needing to do?

/b

On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:

> *User-Agent: eXosip/3.1.0^M
> Content-Length:
>
>
> my question is:
>
> Is it possible to send the call to z.z.z.z , receive the SIP 302 ,
> process the data in Contact field  and redirect to the new destination
> contained in *Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
> *without closing the session.
> i red something about   <action application="set"
> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
>
> Any ideas on this matter  will be highly appreciated.
> Best Regards
> Chav
>

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From brian at freeswitch.org  Sat Dec 13 13:30:40 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Dec 2008 15:30:40 -0600
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
Message-ID: <830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>

I don't get what you're saying.. this looks 100% OK.  Also make sure  
you have the dev tools from apple and not automake, autoconf and  
libtool installed via fink,darwinports or any other such method.  If  
you can provide a bit more information maybe I can see what is wrong.

/b

On Dec 13, 2008, at 3:00 PM, martin joseph wrote:

> I get the following and have issues making it...
>
> Updated to revision 10753.
> mail:/usr/src/freeswitch/trunk root# ./bootstrap.sh
> bootstrap: checking installation...
> bootstrap: autoconf version 2.61 (ok)
> bootstrap: automake version 1.10 (ok)
> bootstrap: libtool version 1.5.24 (ok)
>
> Thanks for the info,
> Marty



From mike at jerris.com  Sat Dec 13 13:51:17 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 13 Dec 2008 16:51:17 -0500
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081213020136.GA16597@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
	<20081213013538.GA16338@jdc.jasonjgw.net>
	<20081213020136.GA16597@jdc.jasonjgw.net>
Message-ID: <661B71A1-D73F-43F0-97BA-11D6798D1486@jerris.com>

Please file a bug on Jira.freeswitch.org on this issue so we make sure  
it gets adressef in the debs.

Mike

On Dec 12, 2008, at 9:01 PM, Jason White <jason at jasonjgw.net> wrote:

> The problem is now solved.
>
> It turned out to be permissions: the freeswitch user wasn't added to  
> the audio
> group in /etc/group, hence didn't have permission to interrogate the  
> audio
> devices.
>
> Perhaps a future version of the Debian package could address this,  
> or at least
> it should be noted somewhere.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mike at jerris.com  Sat Dec 13 13:54:16 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 13 Dec 2008 16:54:16 -0500
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>
References: <49427CC6.2090407@ewetel.de>
	<87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>
Message-ID: <C0C90F5C-CC42-451A-AC91-D35BF4846EAE@jerris.com>

Please file a bug on this issue

Mike

On Dec 13, 2008, at 3:51 PM, "Michael Collins" <msc at freeswitch.org>  
wrote:

> On Fri, Dec 12, 2008 at 7:01 AM, Helmut Kuper  
> <helmut.kuper at ewetel.de> wrote:
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hello,
>>
>> I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
>> AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu  
>> 8.04
>> server in a non-root environment.
>>
>> We experienced a timer problem which led to this FS console error  
>> message:
>>
>> [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries
>>
>>
>> During anylizing this we found that q921 T203 is never reset when  
>> link
>> is in state "Multiple Frame Mode Established" and SABME frames are
>> received by FS. So it must timeout regardless if SABME frames are
>> received or not.
>> Additionally we found that the default T203 value (10 sec) was too  
>> short
>> for AVAYA (it has to be >=19 sec)
>>
>> To fix the problem we changed two things in q921.c:
>>
>> Change T203 default value from 10 sec to 20000 sec
>> Line 406: trunk->T203Timeout = 20000;
>>
>> Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
>> received SABME frame
>> Line 1996: Q921T203TimerReset(trunk, tei);
>>
>> After recompiling FS the Error disapeared. Next week we will do some
>> calls over the link to make sure there are no other side effects.
>>
>> Is it planned to make the q921 timeouts configurable in  
>> openzap.conf or
>> in openzap.conf.xml?
>>
>> best regards
>> Helmut
>>
>>
>> PS: My openzap configs:
>>
>> openzap.conf
>>
>> [span wanpipe PRI_1]
>> trunk_type => E1
>> b-channel => 1:1-15
>> d-channel => 1:16
>> b-channel => 1:17-31
>>
>>
>>
>>
>> openzap.conf.xml
>>
>> <configuration name="openzap.conf" description="OpenZAP  
>> Configuration">
>> <settings>
>>   <param name="debug" value="7"/>
>>   <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>   <!--<param name="enable-analog-option" value="call-swap"/>-->
>>   <!--<param name="enable-analog-option" value="3-way"/>-->
>> </settings>
>>  <pri_spans>
>>    <span name="PRI_1">
>>      <!-- Log Levels: none, alert, crit, err, warning, notice, info,
>> debug -->
>>      <param name="q921loglevel" value="debug"/>
>>      <param name="q931loglevel" value="debug"/>
>>      <param name="mode" value="user"/>
>>      <param name="dialect" value="Oh this is not my dialect"/>
>>      <param name="dialplan" value="XML"/>
>>      <param name="context" value="default"/>
>>    </span>
>>  </pri_spans>
>> </configuration>
>>
>> Very interesting here is, that the "dialect" parameter doesn't seem  
>> to
>> have an effect on FS. I use that one above without any errors or  
>> warning
>> and I guess that was not intended.
>
> At this point in the OZ development we've got it set to default to
> "national" if the dialect isn't otherwise properly specified. It does
> make sense to throw an error if the dialect is not properly specified,
> even if we still default to national.
>
> -MC
>
>>
>>
>>
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.9 (MingW32)
>>
>> iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
>> BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
>> =geXp
>> -----END PGP SIGNATURE-----
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From astmac at stillnewt.org  Sat Dec 13 13:59:19 2008
From: astmac at stillnewt.org (Martin Joseph)
Date: Sat, 13 Dec 2008 13:59:19 -0800
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
	<830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
Message-ID: <52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>


On Dec 13, 2008, at 1:30 PM, Brian West wrote:

> I don't get what you're saying.. this looks 100% OK.  Also make sure
> you have the dev tools from apple and not automake, autoconf and
> libtool installed via fink,darwinports or any other such method.  If
> you can provide a bit more information maybe I can see what is wrong.
Yes, I think it looks good too.  But make fails with the Jira issue  
that has been going on for ages.

I have never used fink or darwinports or any other such methods on  
this box so that's out.

I definitely do have the Apple devtools for 10.4 installed.

I have no problem making the 1.01 FS from the tarball, but as it  
seems you are telling everyone to upgrade to the SVN trunk, I would  
love to do that also.  However, I am frustrated by my inability to  
get that going, as well as a severe lack of time.

Thanks for any help or ideas,
Marty


>
> /b
>
> On Dec 13, 2008, at 3:00 PM, martin joseph wrote:
>
>> I get the following and have issues making it...
>>
>> Updated to revision 10753.
>> mail:/usr/src/freeswitch/trunk root# ./bootstrap.sh
>> bootstrap: checking installation...
>> bootstrap: autoconf version 2.61 (ok)
>> bootstrap: automake version 1.10 (ok)
>> bootstrap: libtool version 1.5.24 (ok)
>>
>> Thanks for the info,
>> Marty
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From brian at freeswitch.org  Sat Dec 13 14:07:07 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Dec 2008 16:07:07 -0600
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
	<830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
	<52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>
Message-ID: <1848C35A-040F-4365-AA17-E43C9B32E11D@freeswitch.org>

Marty,
	Can you point out where its failing?  Nobody has been able to  
reproduce the issue that was reported on jira.  Even Anthony can't and  
he's on 10.4.  I'm on 10.5 and I don't have any issues either.  So if  
you can pin point the exact place where it fails we can look at it  
closer.

/b

On Dec 13, 2008, at 3:59 PM, Martin Joseph wrote:

> Yes, I think it looks good too.  But make fails with the Jira issue
> that has been going on for ages.
>
> I have never used fink or darwinports or any other such methods on
> this box so that's out.
>
> I definitely do have the Apple devtools for 10.4 installed.
>
> I have no problem making the 1.01 FS from the tarball, but as it
> seems you are telling everyone to upgrade to the SVN trunk, I would
> love to do that also.  However, I am frustrated by my inability to
> get that going, as well as a severe lack of time.
>
> Thanks for any help or ideas,
> Marty



From msc at freeswitch.org  Sat Dec 13 14:53:08 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 14:53:08 -0800
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <C0C90F5C-CC42-451A-AC91-D35BF4846EAE@jerris.com>
References: <49427CC6.2090407@ewetel.de>
	<87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>
	<C0C90F5C-CC42-451A-AC91-D35BF4846EAE@jerris.com>
Message-ID: <87f2f3b90812131453q7225c813s8c00a4388e082c40@mail.gmail.com>

Done: http://jira.freeswitch.org/browse/OPENZAP-37
-MC

On Sat, Dec 13, 2008 at 1:54 PM, Michael Jerris <mike at jerris.com> wrote:
> Please file a bug on this issue
>
> Mike
>
> On Dec 13, 2008, at 3:51 PM, "Michael Collins" <msc at freeswitch.org>
> wrote:
>
>> On Fri, Dec 12, 2008 at 7:01 AM, Helmut Kuper
>> <helmut.kuper at ewetel.de> wrote:
>>> -----BEGIN PGP SIGNED MESSAGE-----
>>> Hash: SHA1
>>>
>>> Hello,
>>>
>>> I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
>>> AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu
>>> 8.04
>>> server in a non-root environment.
>>>
>>> We experienced a timer problem which led to this FS console error
>>> message:
>>>
>>> [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries
>>>
>>>
>>> During anylizing this we found that q921 T203 is never reset when
>>> link
>>> is in state "Multiple Frame Mode Established" and SABME frames are
>>> received by FS. So it must timeout regardless if SABME frames are
>>> received or not.
>>> Additionally we found that the default T203 value (10 sec) was too
>>> short
>>> for AVAYA (it has to be >=19 sec)
>>>
>>> To fix the problem we changed two things in q921.c:
>>>
>>> Change T203 default value from 10 sec to 20000 sec
>>> Line 406: trunk->T203Timeout = 20000;
>>>
>>> Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
>>> received SABME frame
>>> Line 1996: Q921T203TimerReset(trunk, tei);
>>>
>>> After recompiling FS the Error disapeared. Next week we will do some
>>> calls over the link to make sure there are no other side effects.
>>>
>>> Is it planned to make the q921 timeouts configurable in
>>> openzap.conf or
>>> in openzap.conf.xml?
>>>
>>> best regards
>>> Helmut
>>>
>>>
>>> PS: My openzap configs:
>>>
>>> openzap.conf
>>>
>>> [span wanpipe PRI_1]
>>> trunk_type => E1
>>> b-channel => 1:1-15
>>> d-channel => 1:16
>>> b-channel => 1:17-31
>>>
>>>
>>>
>>>
>>> openzap.conf.xml
>>>
>>> <configuration name="openzap.conf" description="OpenZAP
>>> Configuration">
>>> <settings>
>>>   <param name="debug" value="7"/>
>>>   <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>   <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>   <!--<param name="enable-analog-option" value="3-way"/>-->
>>> </settings>
>>>  <pri_spans>
>>>    <span name="PRI_1">
>>>      <!-- Log Levels: none, alert, crit, err, warning, notice, info,
>>> debug -->
>>>      <param name="q921loglevel" value="debug"/>
>>>      <param name="q931loglevel" value="debug"/>
>>>      <param name="mode" value="user"/>
>>>      <param name="dialect" value="Oh this is not my dialect"/>
>>>      <param name="dialplan" value="XML"/>
>>>      <param name="context" value="default"/>
>>>    </span>
>>>  </pri_spans>
>>> </configuration>
>>>
>>> Very interesting here is, that the "dialect" parameter doesn't seem
>>> to
>>> have an effect on FS. I use that one above without any errors or
>>> warning
>>> and I guess that was not intended.
>>
>> At this point in the OZ development we've got it set to default to
>> "national" if the dialect isn't otherwise properly specified. It does
>> make sense to throw an error if the dialect is not properly specified,
>> even if we still default to national.
>>
>> -MC
>>
>>>
>>>
>>>
>>> -----BEGIN PGP SIGNATURE-----
>>> Version: GnuPG v1.4.9 (MingW32)
>>>
>>> iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
>>> BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
>>> =geXp
>>> -----END PGP SIGNATURE-----
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Sat Dec 13 14:56:37 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 14:56:37 -0800
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <661B71A1-D73F-43F0-97BA-11D6798D1486@jerris.com>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
	<20081213013538.GA16338@jdc.jasonjgw.net>
	<20081213020136.GA16597@jdc.jasonjgw.net>
	<661B71A1-D73F-43F0-97BA-11D6798D1486@jerris.com>
Message-ID: <87f2f3b90812131456s6bfe35eev8bbc1f04b831cdba@mail.gmail.com>

Done: http://jira.freeswitch.org/browse/FSBUILD-95

On Sat, Dec 13, 2008 at 1:51 PM, Michael Jerris <mike at jerris.com> wrote:
> Please file a bug on Jira.freeswitch.org on this issue so we make sure
> it gets adressef in the debs.
>
> Mike
>
> On Dec 12, 2008, at 9:01 PM, Jason White <jason at jasonjgw.net> wrote:
>
>> The problem is now solved.
>>
>> It turned out to be permissions: the freeswitch user wasn't added to
>> the audio
>> group in /etc/group, hence didn't have permission to interrogate the
>> audio
>> devices.
>>
>> Perhaps a future version of the Debian package could address this,
>> or at least
>> it should be noted somewhere.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Sat Dec 13 15:04:11 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 15:04:11 -0800
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <03c801c95d68$9880ce50$33014c0a@ws4>
References: <87f2f3b90812122239y127f35d6r80cd1286b7dddae8@mail.gmail.com>
	<03c801c95d68$9880ce50$33014c0a@ws4>
Message-ID: <87f2f3b90812131504ue83ddd3qa76a2379f013d531@mail.gmail.com>

On Sat, Dec 13, 2008 at 1:20 PM, Frank @ Impact <frank at impactfax.com> wrote:
> Michael,
>
> Got it working.  Just a little simpler then you outlined.
> I just added to my xml dialplan this line.
>
> <action application="set" data="execute_on_answer=lua uuid_send_dtmf.lua
> ${uuid} 20 123"/>
>
> I added this just before the bridge application.

Nice - simpler is almost always better! :)

>
> I did this instead of adding an extra extension to transfer to on
> answer.  Everything worked well.  The DTMF was played to the calling
> party.  Out of curiosity, if we wanted also to play the DTMF to the
> called party also, what would we have to give uuid_send_dtmf?
> Particularly since we call it before the bridge.
>

Definitely need the uuid of the leg in question. Could you pastebin or
email a sample dialplan? We could probably work it out together.

> Can uuid_send_dtmf accept anything like "w" for wait or anything else
> special for DTMF stuff?

The uuid_send_dtmf api cannot, but you could easily modify or create a
new version of my Lua script that accepts more (or different)
arguments. I suppose the trick there is that you'd need to read up on
Lua, which I highly recommend anyway because if you know Lua then you
can leverage some serious power in your dialplans.

>
> Also, I got an error output to the console when the sched_api was run.
> See below.
> ******
> 2008-12-13 16:07:28 [NOTICE] switch_cpp.cpp:1050 console_log() apicmd:
> sched_api
> 2008-12-13 16:07:28 [NOTICE] switch_cpp.cpp:1050 console_log() apiarg:
> +20 none uuid_send_dtmf 37618e54-c959-11dd-bc73-0923daa880b2 123
>
> 2008-12-13 16:07:28 [ERR] switch_cpp.cpp:1050 console_log() Result is
> +OK Added: 49751
> ******
> is this ERR anything to worry about even though we got a result ok?
>

I believe this "error" is innocuous. Sometimes the devs will log
certain events to the console as ERR so that they stand out during
debugging.
-MC
> Thanks again for the help.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
>
> Frank,
>
> I found a simple way to handle this scenario. I decided just to create
> a small Lua script that would do the job. It's committed in latest
> trunk. Look in src/scripts/contrib/mcollins for uuid_send_dtmf.lua. It
> has comments on how to call it, including a sample dp call.
>
> The way I would use this in your scenario is to setup a destination
> using the execute_on_answer variable.
> http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer
>
> Have the destination be an extension that does something like this:
> <extension...>
>  <condition...>
>    <action application="answer"/>
>    <action application="lua" data="uuid_send_dtmf ${uuid} 10 123"/>
>    ...rest of diaplan...
>  </condition>
> </extension>
>
> The 10 means ten seconds, the 123 means send the dtmf digits 1,2,3 in
> order. You can tinker with the settings as you see fit.
>
> Please let me know how it goes. BTW, be sure to put the Lua script in
> /usr/local/freeswitch/scripts or specify the complete path name when
> calling the lua app in the dialplan.
>
> -MC
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Sat Dec 13 15:40:15 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 15:40:15 -0800
Subject: [Freeswitch-users] Maintaining call detail record
In-Reply-To: <614188.90296.qm@web30706.mail.mud.yahoo.com>
References: <614188.90296.qm@web30706.mail.mud.yahoo.com>
Message-ID: <87f2f3b90812131540v78272fe2m77ed194d07961200@mail.gmail.com>

Faisal,

A few things to keep in mind:

In cdr_csv.conf.xml you need to specify the correct template. There are
several templates specified. I don't know much about the perl script
mentioned there but I don't know if that is how I would approach the
situation personally. One of the templates is called "sql" and it creates
SQL statement for each cdr. You could literally pipe the contents of the cdr
file into MySQL and it will load the data into your table. However you will
need to handle the log file rotation. Some people use cron to send a HUP
signal to the freeswitch process which then rotates the log and Master.csv
files. I recommend you look at the cdr-csv directory before and after a
rotation so that you can see exactly what happens.

Here's a brief checklist for you to help you get going:
1 select the "sql" template in cdr_csv.conf.xml
 <param name="default-template" value="sql"/>

2 create a MySQL database for your cdr data

3 create a table called "cdr" (or rename the table used in the "sql"
template)
 the table needs to have all the fields laid out the way the template lays
them out
 this page can be used as a reference to get you started, but note the these
fields are NOT laid out the same way the sql template lays them out
 http://wiki.freeswitch.org/wiki/Mod_cdr#MySQL_Schema

4 decide how frequently you want to rotate log files and then set up a cron
job that sends the HUP signal:
 kill -hup `cat /usr/local/freeswitch/log/freeswitch.pid`
 After the kill -hup is sent your /usr/local/freeswitch/log/cdr-csv will
look something like this:
 -rw------- 1 root root        0 Dec 13 15:24 Master.csv
 -rw------- 1 root root  1473657 Dec 12 22:13 Master.csv.2008-12-13-15-24-16

  The file Master.csv.YYYY-MM-DD-hh-mm-ss now has the most recent CDR's.

5 run the most recent file through mysql. it is essentially just a text file
with a bunch of INSERT INTO statements
  mysql -u user -p password < Master.csv.YYYY-MM-DD-hh-mm-ss
  rm -f Master.csv.YYYY-MM-DD-hh-mm-ss

steps 4 and 5 could all be in the cron job which just has a script do all
the work. if you need assistance with setting up scripts and doing cron jobs
then i recommend that you manually do the steps one at a time and see
exactly what is happening and then learn how to do the shell script + cron
job.

Good luck!
-MC

P.S. - if anyone already has done all of this and is willing to share
his/her experiences please contact me off list as I would like to talk about
getting a wiki page set up that is dedicated to this sort of thing.

On Sat, Dec 13, 2008 at 1:44 AM, Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>
wrote:
> How can i interface fs with mysql in order to maintain calls record like
> caller id and time n date of call etc. I ve worked on xml cdr but it
> contains too much info, more than i need and in a format which is not
easily
> understandable. I also tried perl coding mentioned on the link at the
bottom
> of the page
> http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r
> displayed during its execution. Is there any easy method for that. Plz
help
> me.
>

> faisal
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Dec 13 15:49:05 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 15:49:05 -0800
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers
In-Reply-To: <05FEA4243A6C422DB5A3D7838AA709FE@test>
References: <1229081927.4100.7.camel@gathern.lan>
	<05FEA4243A6C422DB5A3D7838AA709FE@test>
Message-ID: <87f2f3b90812131549g3ed2e25dg81ec4bea42d497fd@mail.gmail.com>

Also, the other question is this: do you *need* early media? If not then
Darren's suggestion is definitely the way to go. Note that if you ignore
early media then all calls that fail will show up as a NO ANSWER. If this
doesn't work for you then ignoring early media is not an option, in which
case there simply is no perfect way to do it and you just have to make the
best of it. What I've done in the past is something like this:

originate openzap/1/a/5551212 825551212

Then I define an extension that matches on ^82(\d+)$ and does something like
this
<action application="pre_answer"/>
<action application="set" data="execute_on_answer=transfer IVR_ANSWER"/>
<action application="sleep" data="20000"/>
...handle non-answered calls

Then I define another extension that matches on ^IVR_ANSWER$ and does
something like this
<action application="sleep" data="500"/>
<action application="ivr" data="my_ivr"/>
...etc...

The idea for me is to handle the different scenarios I might face when
dialing. At the very least if the call goes unanswered then I have the
hangup_cause variable that tells me if it was busy, no answer, invalid, etc.

Hope that helps.
-MC

On Fri, Dec 12, 2008 at 5:48 AM, Darren Schreiber <d at d-man.org> wrote:

> How are you originating calls? You probably need to add
> {ignore_early_media=true}. This tells FreeSWITCH not to return from
> origination when early media (progress/ringing) was received (I think
> anyway)...
>
> See http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media
>
> There is a sample of this in use with the originate command here:
> http://wiki.freeswitch.org/wiki/Mod_commands#originate (about halfway
> down)
>
> Setting channel variables before doing the originate
>
>   originate {ignore_early_media=true}sofia/
> mydomain.com/18005551212 at 1.2.3.4
> 15555551212
>
>
>
> Since you are making a dialer, you may want to start the originations in
> the
> background and move on to the next call while tweaking the timeout value
> for
> originated calls. From the WIKI again:
>
> "You can originate a call in the background (asynchronously) and playback a
> message with a 60 second timeout.
>
>   bgapi originate
> {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number
> &playback(message)"
>
> - Darren
>
>
>
> -----Original Message-----
> From: Alexandru Nedelcu [mailto:alex at sinapticode.ro]
> Sent: Friday, December 12, 2008 3:39 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Freeswitch streamFile when the user answers
>
> Hi,
>
> I'm working on a simple dialer, and I have the following problem: the audio
> file starts playing before the user answeres the phone (while it's
> ringing).
> It only works when I introduce a delay, but that doesn't seem right.
>
> For instance in the asterisk context referred in the call files, I had:
>
> exten => s,4,Answer
> exten => s,n,Wait(2)
> exten => s,n,Background(${SOUNDFILE})
> And indeed it played a soundfile 2 seconds after the called person picked
> up
> the phone
>
> In FS I currently initiate calls like this:
>
> session.waitForAnswer(10000);
>
> if (session.ready()) {
>    session.sleep(2000);
>    session.streamFile(/*...*/);
> }
>
> Is this right?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Sat Dec 13 15:52:59 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 15:52:59 -0800
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <op.ulwl5dbessieo1@woof.us.nortel.com>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
Message-ID: <87f2f3b90812131552p295eb1c4gee7d29f7a72624dc@mail.gmail.com>

Another option for you, ironically, is to have the freeswitch.log file plus
the other log files that are not freeswitch's, to go into a third directory
that is uniquely set up for this purpose. That way it wouldn't be disruptive
to move a bunch of files from log to db. All you'd have to do is modify the
logfile.conf.xml file and pick a new path for your freeswitch.log file...

-MC

On Tue, Dec 9, 2008 at 8:45 AM, Andy Spitzer <woof at nortel.com> wrote:

> Woof!
>
> It appears that FreeSWITCH writes
>
>   freeswitch.history
>   freeswitch.log
>   freeswitch.pid
>   freeswitch.xml.fsxml
>
> to the -log directory.
>
> Is there a way to put the files other than freeswitch.log into the -db
> directory instead?
>
> In my environment we archive and rotate everything in the log directory
> (which includes logs beside FreeSWITCH's), and these other FreeSWITCH files
> are getting rotated.  Yeah, I can explicitly exclude them, but to me it
> seems those really belong in the -db directory anyway, as they are
> inherently data needed for the current executable of FreeSWITCH, and not
> logs.
>
> --Woof!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From astmac at stillnewt.org  Sat Dec 13 16:23:44 2008
From: astmac at stillnewt.org (martin joseph)
Date: Sat, 13 Dec 2008 16:23:44 -0800
Subject: [Freeswitch-users] ./configure fails on 10.4.11
In-Reply-To: <1848C35A-040F-4365-AA17-E43C9B32E11D@freeswitch.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
	<830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
	<52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>
	<1848C35A-040F-4365-AA17-E43C9B32E11D@freeswitch.org>
Message-ID: <B9DF8866-2120-4E5C-8FFB-352A11A71438@stillnewt.org>


On Dec 13, 2008, at 2:07 PM, Brian West wrote:

> Marty,
> 	Can you point out where its failing?
./configure fails as follows (just as in the JIRA)

checking for a BSD-compatible install... /usr/bin/install -c
./configure: line 4112: syntax error near unexpected token  
`build_libtool_libs,'
./configure: line 4112: `    _LT_DECL(build_libtool_libs,  
enable_shared, 0,'
configure: error: ./configure.gnu failed for libs/openmrcp

> Nobody has been able to
> reproduce the issue that was reported on jira.  Even Anthony can't and
> he's on 10.4.
Huh,  I would love to figure this out, as it seems certain to be  
somehow specific to my install.
>   I'm on 10.5 and I don't have any issues either.  So if
> you can pin point the exact place where it fails we can look at it
> closer.
I would love that.

Thanks again for your excellent software and help.
Marty

>
> /b
>
> On Dec 13, 2008, at 3:59 PM, Martin Joseph wrote:
>
>> Yes, I think it looks good too.  But make fails with the Jira issue
>> that has been going on for ages.
>>
>> I have never used fink or darwinports or any other such methods on
>> this box so that's out.
>>
>> I definitely do have the Apple devtools for 10.4 installed.
>>
>> I have no problem making the 1.01 FS from the tarball, but as it
>> seems you are telling everyone to upgrade to the SVN trunk, I would
>> love to do that also.  However, I am frustrated by my inability to
>> get that going, as well as a severe lack of time.
>>
>> Thanks for any help or ideas,
>> Marty
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From chavpaskov at shaw.ca  Sat Dec 13 16:38:41 2008
From: chavpaskov at shaw.ca (Chav Paskov)
Date: Sat, 13 Dec 2008 16:38:41 -0800
Subject: [Freeswitch-users] how to handle  returned sip 302 dialplan
In-Reply-To: <9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
Message-ID: <49445591.90505@shaw.ca>

Brian West wrote:
> Chav,
> Once the 302 is received by FreeSWITCH it will follow it to the 
> contact listed in the 302.  What else are you needing to do?
>
> /b
>
> On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
>
>> *User-Agent: eXosip/3.1.0^M
>> Content-Length:  
>>
>>
>> my question is:
>>
>> Is it possible to send the call to z.z.z.z , receive the SIP 302 , 
>> process the data in Contact field  and redirect to the new destination 
>> contained in *Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
>> *without closing the session.
>> i red something about   <action application="set" 
>> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
>>
>> Any ideas on this matter  will be highly appreciated.
>> Best Regards
>> Chav
>>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
Thanks Brian,

probably i should have explained it in more details.
this whole thing started as an attempt to  implement lata ocn /local 
number portability/ instead of  pure per destination routing.
At the moment i have a access to  a service provider  who does 
"dipping"  and returns  the  LATA OCN  data associated with  any dialed 
destination number. it is returned as Contact:  and Content-length:  
fields in 302 message.

in other words:

1. i'm sending to this provider let say - 2025556666  as a destination 
number.
2. they do the dipping  and will return to me  either the new dest #  if 
2025556666 has been ported or if it was not
in content-length field they'll send  lata, ocn  and state and 10 digits 
number.
3. once received i have to  compare the received lata , ocn and state  
date with  a compiled  rate deck / blended from 5 different vendors/  
and pick the lowest rate - effectively building  LCR based on LATA OCN 
STATE info.

Hope this will help to clear the picture.
Regards
Chav




From frank at impactfax.com  Sat Dec 13 20:42:11 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 13 Dec 2008 23:42:11 -0500
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <87f2f3b90812131504ue83ddd3qa76a2379f013d531@mail.gmail.com>
Message-ID: <002701c95da6$54acb9d0$33014c0a@ws4>

Pretty simple actually...

<?xml version="1.0"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="Regex/XML Dialplan">
    <context name="default">
      <extension name="myextension">
        <condition>
	  <action application="enum" data="15555551212"/>
	  <action application="set" data="execute_on_answer=lua
uuid_send_dtmf.lua ${uuid} 10 55"/>
	  <!-- use this if to wrap just sections you want to test -->
	  <action application="tone_detect" data="test 941,1336 rw
+100000 hangup normal_clearing"/>
	  <action application="bridge" data="${enum_auto_route}"/> 
	  <action application="hangup"/>
        </condition>
      </extension>
    </context>
  </section>
</document>

BTW, this darn tone_detect is something I never could get working.  It
did not matter which side I sent the tone from, it never got trapped by
my test here.  The call never hung up on the tone, a 0.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Saturday, December 13, 2008 6:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] schedule a DTMF tone into bridge

>
> I did this instead of adding an extra extension to transfer to on
> answer.  Everything worked well.  The DTMF was played to the calling
> party.  Out of curiosity, if we wanted also to play the DTMF to the
> called party also, what would we have to give uuid_send_dtmf?
> Particularly since we call it before the bridge.
>

Definitely need the uuid of the leg in question. Could you pastebin or
email a sample dialplan? We could probably work it out together.





From jason at jasonjgw.net  Sat Dec 13 22:23:03 2008
From: jason at jasonjgw.net (Jason White)
Date: Sun, 14 Dec 2008 17:23:03 +1100
Subject: [Freeswitch-users] Sip profiles used in bridge application
Message-ID: <20081214062303.GA22331@jdc.jasonjgw.net>

The Wiki page at
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
gives the following example:

<action application="bridge"
data="sofia/sip/9998881111 at sip.yourprovider.com"/>

which, for me at least, doesn't work due to an invalid profile error:

[ERR] mod_sofia.c:2404 sofia_outgoing_channel() Invalid Profile

However, if I replace "sip" in the above action with "external" to specify the
external profile, it works.

I suspect this is a documentation error rather than a code bug. However, I
couldn't find an explanation of the syntax, in particular, whether the profile
name has to match a profile defined in the SIP configuration or whether there
are profile names defined in the code that have special meanings as well.

I'm using the default configuration for now.

I could go through the code to find this out, of course, but I thought it
better to ask here instead.



From msc at freeswitch.org  Sat Dec 13 22:39:55 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 13 Dec 2008 22:39:55 -0800
Subject: [Freeswitch-users] Sip profiles used in bridge application
In-Reply-To: <20081214062303.GA22331@jdc.jasonjgw.net>
References: <20081214062303.GA22331@jdc.jasonjgw.net>
Message-ID: <C1CD13F2-C731-4862-BFE7-0D71003592CC@freeswitch.org>

Jason,

Thanks for pointing this out. You are correct. This is a case of  
development moving faster than documentation efforts. I will update  
the wiki.

-MC

Sent from my iPhone

On Dec 13, 2008, at 10:23 PM, Jason White <jason at jasonjgw.net> wrote:

> The Wiki page at
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
> gives the following example:
>
> <action application="bridge"
> data="sofia/sip/9998881111 at sip.yourprovider.com"/>
>
> which, for me at least, doesn't work due to an invalid profile error:
>
> [ERR] mod_sofia.c:2404 sofia_outgoing_channel() Invalid Profile
>
> However, if I replace "sip" in the above action with "external" to  
> specify the
> external profile, it works.
>
> I suspect this is a documentation error rather than a code bug.  
> However, I
> couldn't find an explanation of the syntax, in particular, whether  
> the profile
> name has to match a profile defined in the SIP configuration or  
> whether there
> are profile names defined in the code that have special meanings as  
> well.
>
> I'm using the default configuration for now.
>
> I could go through the code to find this out, of course, but I  
> thought it
> better to ask here instead.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From pabx_freeswitch at telenet.be  Sun Dec 14 04:52:17 2008
From: pabx_freeswitch at telenet.be (henkoegema)
Date: Sun, 14 Dec 2008 04:52:17 -0800 (PST)
Subject: [Freeswitch-users]  libtool version.
Message-ID: <21000027.post@talk.nabble.com>


root at MSI:/home/henkoegema/freeswitch# ./bootstrap.sh 
bootstrap: checking installation...
bootstrap: autoconf version 2.61 (ok)
bootstrap: automake version 1.10.1 (ok)
bootstrap: libtool version 2.2.4 found.    
           You need libtool version 1.5.14 or newer installed     
<------Isn't that what I have  :confused:
           to build FreeSWITCH from SVN.
root at MSI:/home/henkoegema/freeswitch# 

-- 
View this message in context: http://www.nabble.com/libtool-version.-tp21000027p21000027.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From jbr at consiglia.dk  Sun Dec 14 08:08:54 2008
From: jbr at consiglia.dk (Jon Bruel)
Date: Sun, 14 Dec 2008 17:08:54 +0100
Subject: [Freeswitch-users] How to control to domain used in INVITE From
	header
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE02838C515F@SBS2008SERVER.consiglia.local>

The situation is as follows: An incoming call is processed by the FS and sent out to a sip client. I want to control the From header in this outgoing INVITE.
I have tried to set various channel variables, including sip_h_From, in order to control to domain used in the INVITE From header, which for instance looks like this:
From: "JBS (Soft)" <sip:205 at X.X.X.X>
Instead of the server IP address, X.X.X.X, I want to set the SIP-domain used for the specific customer in a multi tenant setup. This from header is used by the phone telephone number list register (at least for the Snom phones), so controlling it is important. How is it done? /Jon
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From mike at jerris.com  Sun Dec 14 08:21:11 2008
From: mike at jerris.com (Michael Jerris)
Date: Sun, 14 Dec 2008 11:21:11 -0500
Subject: [Freeswitch-users] libtool version.
In-Reply-To: <21000027.post@talk.nabble.com>
References: <21000027.post@talk.nabble.com>
Message-ID: <08AE93ED-54E6-4179-BEEB-DF550DD05B5E@jerris.com>

http://jira.freeswitch.org/browse/FSBUILD-82

On Dec 14, 2008, at 7:52 AM, henkoegema wrote:

>
> root at MSI:/home/henkoegema/freeswitch# ./bootstrap.sh
> bootstrap: checking installation...
> bootstrap: autoconf version 2.61 (ok)
> bootstrap: automake version 1.10.1 (ok)
> bootstrap: libtool version 2.2.4 found.
>           You need libtool version 1.5.14 or newer installed
> <------Isn't that what I have  :confused:
>           to build FreeSWITCH from SVN.
> root at MSI:/home/henkoegema/freeswitch#
>
> -- 
> View this message in context: http://www.nabble.com/libtool-version.-tp21000027p21000027.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Dec 14 10:00:18 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Dec 2008 12:00:18 -0600
Subject: [Freeswitch-users] How to control to domain used in INVITE From
	header
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE02838C515F@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE02838C515F@SBS2008SERVER.consiglia.local>
Message-ID: <704CEFC8-9E10-4690-AB8F-942D8382DF9E@freeswitch.org>

the sip_invite_domain variable.

/b

On Dec 14, 2008, at 10:08 AM, Jon Bruel wrote:

> The situation is as follows: An incoming call is processed by the FS  
> and sent out to a sip client. I want to control the >From header in  
> this outgoing INVITE.
>
> I have tried to set various channel variables, including sip_h_From,  
> in order to control to domain used in the INVITE From header, which  
> for instance looks like this:
>
> From: "JBS (Soft)" <sip:205 at X.X.X.X>
>
> Instead of the server IP address, X.X.X.X, I want to set the SIP- 
> domain used for the specific customer in a multi tenant setup. This  
> from header is used by the phone telephone number list register (at  
> least for the Snom phones), so controlling it is important. How is  
> it done? /Jon
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Dec 14 10:01:21 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Dec 2008 12:01:21 -0600
Subject: [Freeswitch-users] Sip profiles used in bridge application
In-Reply-To: <20081214062303.GA22331@jdc.jasonjgw.net>
References: <20081214062303.GA22331@jdc.jasonjgw.net>
Message-ID: <9B569932-48D8-4F66-9CDC-ECBEF7A6C459@freeswitch.org>

Well if the config had a profile called "sip" it would be perfectly  
fine.  Remember the profile names are not set in stone and you can  
name them what ever you wish.

/b

On Dec 14, 2008, at 12:23 AM, Jason White wrote:

>
> [ERR] mod_sofia.c:2404 sofia_outgoing_channel() Invalid Profile



From jason at jasonjgw.net  Sun Dec 14 14:45:35 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 09:45:35 +1100
Subject: [Freeswitch-users] references to source files in error messages
Message-ID: <20081214224535.GA5335@jdc.jasonjgw.net>

I would just like to thank the FreeSWITCH developers for including the source
file names and line numbers in error messages. This is not only helpful to the
authors of the software, but to anyone who can read C code.



From jason at jasonjgw.net  Sun Dec 14 20:41:13 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 15:41:13 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
Message-ID: <20081215044113.GA9555@jdc.jasonjgw.net>

I'm trying to make an outbound call to an IPv6 host using the pre-supplied
internal-ipv6 profile.

<action application="bridge"
data="sofia/internal-ipv6/nnnn at ipv6-host.domain"/>

I can ping the host with ping6, and it has a DNS AAAA record. (There is no A
record as the host is a friend's box behind a NAT).

My own box is also behind a NAT, but it has IPv6 connectivity via a tunnel
broker.

The error in the freeswitch.log file is:
[ERR] sofia_reg.c:1344 sofia_reg_handle_sip_r_challenge() No Matching gateway
found

It's quite possible I could be doing something wrong, or there might be a bug
somewhere, or the provided internal-ipv6 profile might need some adjusting...

Suggestions welcome.



From jason at jasonjgw.net  Sun Dec 14 21:25:18 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 16:25:18 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215044113.GA9555@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
Message-ID: <20081215052518.GA9967@jdc.jasonjgw.net>

Incidentally, if I try it with the external profile, or with my own
external-ipv6 profile (the same as the supplied one, but binding to IPv6
rather than IPv4 addresses), then I get

2008-12-15 16:16:50 [DEBUG] mod_sofia.c:253 sofia_on_hangup()
sofia/external-ipv6/nnnn at ipv6-host.domain Overriding SIP cause 503 with 503
from the other leg

The other party (with a configured FreeSWITCH listening at the IPv6 address in
question) gets the same error upon attempting to call my machine.



From faisalmaqsoodi at yahoo.com  Sun Dec 14 21:43:12 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sun, 14 Dec 2008 21:43:12 -0800 (PST)
Subject: [Freeswitch-users] Maintaining call detail record
Message-ID: <627709.35367.qm@web30703.mail.mud.yahoo.com>

                        
                        

if (typeof YAHOO == "undefined") {
 var YAHOO = {};
}
YAHOO.Shortcuts = YAHOO.Shortcuts || {};
YAHOO.Shortcuts.hasSensitiveText = false;
YAHOO.Shortcuts.sensitivityType = [];
YAHOO.Shortcuts.doUlt = false;
YAHOO.Shortcuts.location = "us";
YAHOO.Shortcuts.document_id = 0;
YAHOO.Shortcuts.document_type = "";
YAHOO.Shortcuts.document_title = "[Freeswitch-users] Maintaining call detail record";
YAHOO.Shortcuts.document_publish_date = "";
YAHOO.Shortcuts.document_author = "faisalmaqsoodi at yahoo.com";
YAHOO.Shortcuts.document_url = "";
YAHOO.Shortcuts.document_tags = "";
YAHOO.Shortcuts.document_language = "english";
YAHOO.Shortcuts.annotationSet = {
"lw_1229319648_0": {
"text": "caller id",
"extended": 0,
"startchar": 188,
"endchar": 196,
"start": 188,
"end": 196,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 0.2054,
"relScore": 5.33095,
"type": ["shortcuts:/concept"],
"category": ["CONCEPT"],
"wikiId": "Caller_ID",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked",
"metaData": {
"visible": "true"
}
 },
"lw_1229319648_1": {
"text": "http://wiki.freeswitch.org/wiki/Mod_cdr_csv,",
"extended": 0,
"startchar": 426,
"endchar": 469,
"start": 426,
"end": 469,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 1,
"relScore": 0,
"type": ["shortcuts:/us/instance/identifier/URL"],
"category": ["IDENTIFIER"],
"wikiId": "",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "",
"metaData": {
"visible": "true"
}
 }
};
YAHOO.Shortcuts.headerID = "284c8f98b4fb0aebc968053934caa66b";

How can i interface fs with mysql in order to maintain calls record like caller id
and time n date of call etc. I ve worked on xml cdr but it contains too
much info, more than i need and in a format which is not easily
understandable. I also tried perl coding mentioned on the link at the
bottom of the page
http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r displayed during its execution. Is there any easy method for that. Plz help me.
 
?????????????????????????????????????????????????????????????????????????????????????? faisal



      
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From helmut.kuper at ewetel.de  Sun Dec 14 23:24:57 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 15 Dec 2008 08:24:57 +0100
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <87f2f3b90812131453q7225c813s8c00a4388e082c40@mail.gmail.com>
References: <49427CC6.2090407@ewetel.de>	<87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>	<C0C90F5C-CC42-451A-AC91-D35BF4846EAE@jerris.com>
	<87f2f3b90812131453q7225c813s8c00a4388e082c40@mail.gmail.com>
Message-ID: <49460649.6030302@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1


Hello Michael 1+2,

thank you for opening a bug. Otherwise I would have opened a bug today
after testing it a bit more.

regards
Helmut

Am 13.12.2008 23:53, schrieb Michael Collins:
> Done: http://jira.freeswitch.org/browse/OPENZAP-37
> -MC
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RdcAnRasalavYRJ9hRcj5DjWYooZS6vb
=cPz3
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From helmut.kuper at ewetel.de  Sun Dec 14 23:35:17 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 15 Dec 2008 08:35:17 +0100
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <191c3a030812120718n7d8c5410y2ad3cebab8f5be3b@mail.gmail.com>
References: <49427CC6.2090407@ewetel.de>
	<191c3a030812120718n7d8c5410y2ad3cebab8f5be3b@mail.gmail.com>
Message-ID: <494608B5.4060305@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

OK, I opened a bug for T203 on jira:
http://jira.freeswitch.org/browse/OPENZAP-38

regards
helmut

Am 12.12.2008 16:18, schrieb Anthony Minessale:
> if you open a jira issue on it we can probably add your patch and/or the
> config option.
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59QAoKlallmRwdKBuTOUJcVZMDgQL0bU
=idOI
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From stevecrozz at gmail.com  Sun Dec 14 23:32:08 2008
From: stevecrozz at gmail.com (Stephen Crosby)
Date: Sun, 14 Dec 2008 23:32:08 -0800
Subject: [Freeswitch-users] running custom script with bind_meta_app
Message-ID: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>

I'm Stephen Crosby, and I've just started working with freeswitch.
It's been great so far.

I want to run a custom script inside a conference when a DTMF sequence
is entered. I found bind_meta_app and thought it would be perfect, but
I can't seem to get it to work. When I dial-in and press *8, I get no
debugging output at all. When I press another sequence like *9 for
instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf()
sofia/external/5593495805 at sip.gafachi.com Ignoring meta digit '9' not
mapped. The script I wrote has been tested with "jsrun script.js" from
the command line and it does work. I've got the debugging level all
the way up and there's just not much for me to go on. Any help would
be greatly appreciated.

<extension name="gafachi-DID1">
  <condition field="destination_number" expression="^xxxxxxxxxxx$">
    <action application="answer"/>
    <action application="bind_meta_app" data="8 ab s
javascript::$${base_dir}/scripts/script.js"/>
    <action application="conference" data="3000 at default"/>
  </condition>
</extension>

--Stephen


From jason at jasonjgw.net  Mon Dec 15 00:01:05 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 19:01:05 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215052518.GA9967@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
Message-ID: <20081215080105.GA11352@jdc.jasonjgw.net>

Turning on all of the Sofia debugging options reveals the following.

FS sends out the invite (UDP, 1245 bytes).

Then we get:

nta: timer shortened to 500 ms
tport_wakeup_pri(0x2490700): events ERR
tport_udp_error: icmp(6) message was truncated (at 832)
tport_udp_error: Message too long (90) [icmp6 type=2 code=0 info=00000500]
	reported by [2001:470:0:5d::2]:65535
tport_release(0x2490700): 0x7fe244098c10 by 0x25022a0 with (nil)
nta: INVITE (108544297): Message too long (90) with udp/[xxxxx - remote host's
IPv6 address]:5060
nua(0x7fe24409af90): event r_invite 503 Service Unavailable
nua(0x7fe24409af90): call state changed: calling -> init
nua: nua_application_event: entering
nua(0x7fe24409af90): event i_state 503 Service Unavailable
nua(0x7fe24409af90): event i_terminated 503 Service Unavailable
nua: nua_handle_magic: entering
nua(0x7fe24409af90): removing session usage
nua: nua_application_event: entering
nta_leg_destroy(0x7fe2440822f0)
2008-12-15 18:14:27 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/external-ipv6/nnnn at remote-host.domain entering state [terminated]

The IPv6 address that sent the "message too long" error is presumably that of
a router between my FreeSWITCH box and my friend's FreeSWITCH box.

Is there a way around this?



From jan.kubr at gmail.com  Mon Dec 15 01:27:25 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Mon, 15 Dec 2008 10:27:25 +0100
Subject: [Freeswitch-users] Interrupting read application with DTMF
Message-ID: <698401620812150127s4f96a7c4h3516da007f7399a0@mail.gmail.com>

Hi,
I have been having some troubles with the read application for quite a
while which I haven't been able to solve yet.
I have Freeswitch connected to a SIP gateway to accept calls from a
landline-like number. For the incoming calls I have a simple testing
dialplan:

<action application="read" data="1 1
en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
<action application="phrase" data="spell,${res}"/>

The behavior I have a problem with is that the read app should
terminate when I press a digit and the execution should jump to the
next action - meaning the playback of the file should be interrupted.
The problem is that when I call the public number from my cell phone
this works only about 50% of the time. In the other cases I need to
wait for the wav file to be played (or press the digit two or three
times). When using a SIP phone it always works.

Today I tried to convert the wav file the read app plays to the GSM
format and found out it fixed the problem! Now I can almost always
interrupt the read app with DTMF from my cell phone. Doing the same
from my SIP phone doesn't work well though when the file is GSM.

Can someone explain me what is going on here and what is the right
approach? I'm on revision 10751. I've tried to set a few configuration
variables based on suggestions from this list, but it didn't make any
difference.

Thanks,
Jan Kubr


From jonas.gauffin at gmail.com  Mon Dec 15 02:19:44 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 15 Dec 2008 11:19:44 +0100
Subject: [Freeswitch-users] Bridging through gateway
Message-ID: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>

I'm trying to bridge using a non-registered gateway. And I
get MANDATORY_IE_MISSING back. Why is that?

2008-12-15 11:06:49 [DEBUG] switch_core_state_machine.c:152
switch_core_standard_on_execute()
sofia/internal/u1000044 at 192.168.1.112:5070Execute
bridge(sofia/default/
0236661201 at sip-corporate2.tele2.se)
[.....]
2008-12-15 11:06:49 [DEBUG] sofia.c:2511 sofia_handle_sip_i_state() Channel
sofia/internal/0236661201 at sip-corporate2.tele2.se entering state [calling]
2008-12-15 11:06:49 [ERR] sofia_reg.c:1312
sofia_reg_handle_sip_r_challenge() No Matching gateway found
2008-12-15 11:06:49 [NOTICE] sofia_reg.c:1333
sofia_reg_handle_sip_r_challenge() Hangup sofia/internal/
0236661201 at sip-corporate2.tele2.se [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING]
2008-12-15 11:06:49 [DEBUG] switch_channel.c:1478
switch_channel_perform_hangup() Send signal sofia/internal/
0236661201 at sip-corporate2.tele2.se [KILL]
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From FranziskaRoehler at aol.com  Mon Dec 15 01:57:57 2008
From: FranziskaRoehler at aol.com (=?iso-8859-1?Q?Franziska_R=F6hler?=)
Date: Mon, 15 Dec 2008 10:57:57 +0100
Subject: [Freeswitch-users] Openzap ERROR can't dial
Message-ID: <000c01c95e9b$9cbaf610$6445310a@Franzi>

Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

 

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN). But when I make an outbound call, I have an error. FS
can?t dial.

 

[ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING]

 

I have this warnings too, when no call is done:

 

2008-12-15 10:11:14 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6

2008-12-15 10:11:15 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 7

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:1
(1:1) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:2
(1:2) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:3
(1:3) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:4
(1:4) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:5
(1:5) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:6
(1:6) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:7
(1:7) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:8
(1:8) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:9
(1:9) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:10
(1:10) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:11
(1:11) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:12
(1:12) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:13
(1:13) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:14
(1:14) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:15
(1:15) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:16
(1:17) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:17
(1:18) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:18
(1:19) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:19
(1:20) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:20
(1:21) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:21
(1:22) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:22
(1:23) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:23
(1:24) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:24
(1:25) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:25
(1:26) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:26
(1:27) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:27
(1:28) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:28
(1:29) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:29
(1:30) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:30
(1:31) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:31
(1:16) has alarms [YELLOW]

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:1

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:2

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:3

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:4

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:5

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:6

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:7

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:8

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:9

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:10

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:11

 

What do I wrong?

I hope you can help me!

 

Here my configuration:

 

OPENZAP.CONF

 

[span zt]

name => OpenZap

number => 1

trunk_type => e1

b-channel => 1-15, 17-31

d-channel => 16

 

OPENZAP.CONF.XML

 

<configuration name="openzap.conf" description="OpenZAP Configuration">

  <settings>

    <param name="debug" value="0"/>

  </settings>

  <pri_spans>

    <span id="1">

      <param name="mode" value="user"/>

      <param name="dialect" value="euro"/>

      <param name="dialplan" value="XML"/>

      <param name="context" value="default"/>

    </span>

  </pri_spans>

</configuration>

 

ZT.CONF

 

[defaults]

codec_ms => 20

wink_ms => 150

flash_ms => 750

echo_cancel_level => 64

 

DEFAULT.XML

 

   <extension name="outgoing_pri">

      <condition field="destination_number"
expression="^([0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9])$">

            <action application="set" data="dialed_ext=$1"/>

            <action application="bridge" data="openzap/1/A/${dialed_ext}"/>

      </condition>

    </extension>

 

When I have forget to display some Configuration to , you can tell me!
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From jason at jasonjgw.net  Mon Dec 15 03:03:46 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 22:03:46 +1100
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
Message-ID: <20081215110346.GA12681@jdc.jasonjgw.net>

On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
> I'm trying to bridge using a non-registered gateway. And I
> get MANDATORY_IE_MISSING back. Why is that?

Does the gateway allow unauthenticated clients to make calls? If you obtain a
SIP trace, you'll be able to see whether it's an authentication issue.

As an aside, it would be an improvement to FreeSWITCH if Sofia debugging could
be turned on and off within a running FreeSWITCH instance, including SIP
traces, instead of the administrator's having to restart FreeSWITCH with
environment variables exported, as is presently required according to the
wiki.



From brian at freeswitch.org  Mon Dec 15 06:45:49 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 08:45:49 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215080105.GA11352@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
Message-ID: <38E75A1D-7391-46A8-BD9C-1C851A019625@freeswitch.org>

Are you using SVN trunk?  This has been fixed already as far as I  
remember!!

/b

Sent from my iPhne

On Dec 15, 2008, at 2:01 AM, Jason White <jason at jasonjgw.net> wrote:

> Turning on all of the Sofia debugging options reveals the following.
>
> FS sends out the invite (UDP, 1245 bytes).
>
> Then we get:
>
> nta: timer shortened to 500 ms
> tport_wakeup_pri(0x2490700): events ERR
> tport_udp_error: icmp(6) message was truncated (at 832)
> tport_udp_error: Message too long (90) [icmp6 type=2 code=0  
> info=00000500]
>    reported by [2001:470:0:5d::2]:65535
> tport_release(0x2490700): 0x7fe244098c10 by 0x25022a0 with (nil)
> nta: INVITE (108544297): Message too long (90) with udp/[xxxxx -  
> remote host's
> IPv6 address]:5060
> nua(0x7fe24409af90): event r_invite 503 Service Unavailable
> nua(0x7fe24409af90): call state changed: calling -> init
> nua: nua_application_event: entering
> nua(0x7fe24409af90): event i_state 503 Service Unavailable
> nua(0x7fe24409af90): event i_terminated 503 Service Unavailable
> nua: nua_handle_magic: entering
> nua(0x7fe24409af90): removing session usage
> nua: nua_application_event: entering
> nta_leg_destroy(0x7fe2440822f0)
> 2008-12-15 18:14:27 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state()  
> Channel sofia/external-ipv6/nnnn at remote-host.domain entering state  
> [terminated]
>
> The IPv6 address that sent the "message too long" error is  
> presumably that of
> a router between my FreeSWITCH box and my friend's FreeSWITCH box.
>
> Is there a way around this?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Mon Dec 15 07:05:49 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 09:05:49 -0600
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <20081215110346.GA12681@jdc.jasonjgw.net>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
	<20081215110346.GA12681@jdc.jasonjgw.net>
Message-ID: <432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>

If you don't need auth you don't need a gateway.  sofia/profile/ 
number at remoteip is all you should need.

/b

On Dec 15, 2008, at 5:03 AM, Jason White wrote:

> On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
>> I'm trying to bridge using a non-registered gateway. And I
>> get MANDATORY_IE_MISSING back. Why is that?
>
> Does the gateway allow unauthenticated clients to make calls? If you  
> obtain a
> SIP trace, you'll be able to see whether it's an authentication issue.
>
> As an aside, it would be an improvement to FreeSWITCH if Sofia  
> debugging could
> be turned on and off within a running FreeSWITCH instance, including  
> SIP
> traces, instead of the administrator's having to restart FreeSWITCH  
> with
> environment variables exported, as is presently required according  
> to the
> wiki.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Dec 15 07:08:56 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 09:08:56 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215080105.GA11352@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
Message-ID: <081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>

 From the past emails and the data you have provided so far it makes  
me believe you're not on SVN trunk.  Also the second email looks like  
the far end tries to challenge you and we can't find a matching  
gateway.   I have tested IPv6 with Snom and FS to FS pretty much  
daily.   Can you verify you're on SVN trunk and not 1.0.1?

/b

On Dec 15, 2008, at 2:01 AM, Jason White wrote:

> Turning on all of the Sofia debugging options reveals the following.
>
> FS sends out the invite (UDP, 1245 bytes).
>
> Then we get:
>
> nta: timer shortened to 500 ms
> tport_wakeup_pri(0x2490700): events ERR
> tport_udp_error: icmp(6) message was truncated (at 832)
> tport_udp_error: Message too long (90) [icmp6 type=2 code=0  
> info=00000500]
> 	reported by [2001:470:0:5d::2]:65535
> tport_release(0x2490700): 0x7fe244098c10 by 0x25022a0 with (nil)
> nta: INVITE (108544297): Message too long (90) with udp/[xxxxx -  
> remote host's
> IPv6 address]:5060
> nua(0x7fe24409af90): event r_invite 503 Service Unavailable
> nua(0x7fe24409af90): call state changed: calling -> init
> nua: nua_application_event: entering
> nua(0x7fe24409af90): event i_state 503 Service Unavailable
> nua(0x7fe24409af90): event i_terminated 503 Service Unavailable
> nua: nua_handle_magic: entering
> nua(0x7fe24409af90): removing session usage
> nua: nua_application_event: entering
> nta_leg_destroy(0x7fe2440822f0)
> 2008-12-15 18:14:27 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state()  
> Channel sofia/external-ipv6/nnnn at remote-host.domain entering state  
> [terminated]
>
> The IPv6 address that sent the "message too long" error is  
> presumably that of
> a router between my FreeSWITCH box and my friend's FreeSWITCH box.
>
> Is there a way around this?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Dec 15 07:09:13 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 09:09:13 -0600
Subject: [Freeswitch-users] running custom script with bind_meta_app
In-Reply-To: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>
References: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>
Message-ID: <910AFB7A-9CDE-49AE-A72D-7942F8F79DA3@freeswitch.org>

What are you wanting to accomplish first?

/b

On Dec 15, 2008, at 1:32 AM, Stephen Crosby wrote:

> I'm Stephen Crosby, and I've just started working with freeswitch.
> It's been great so far.
>
> I want to run a custom script inside a conference when a DTMF sequence
> is entered. I found bind_meta_app and thought it would be perfect, but
> I can't seem to get it to work. When I dial-in and press *8, I get no
> debugging output at all. When I press another sequence like *9 for
> instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf()
> sofia/external/5593495805 at sip.gafachi.com Ignoring meta digit '9' not
> mapped. The script I wrote has been tested with "jsrun script.js" from
> the command line and it does work. I've got the debugging level all
> the way up and there's just not much for me to go on. Any help would
> be greatly appreciated.
>
> <extension name="gafachi-DID1">
>  <condition field="destination_number" expression="^xxxxxxxxxxx$">
>    <action application="answer"/>
>    <action application="bind_meta_app" data="8 ab s
> javascript::$${base_dir}/scripts/script.js"/>
>    <action application="conference" data="3000 at default"/>
>  </condition>
> </extension>
>
> --Stephen

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From freeswitch at ptmm.com  Mon Dec 15 02:56:12 2008
From: freeswitch at ptmm.com (Clifford)
Date: Mon, 15 Dec 2008 03:56:12 -0700
Subject: [Freeswitch-users] Multiple mod_portaudio channels
Message-ID: <mailman.0.1229355809.18010.freeswitch-users@lists.freeswitch.org>

I need to set up a system that would allow multiple mod_portaudio channels
to be configured.  
 
The calls would come in over SIP and any calls would be assigned a specific
port on the multi-port sound card.
 
I would probably set it up so specific extensions go to specific ports, for
example 1001 would be port 1 and 1008 would be port 8 on the sound card.
 
The documentation on the mod_portaudio module is very slim.  In fact I would
say it is nearly undocumented.  Is it capable of handling multiple audio
device ports (such as an 8-port LYNX sound card)? 
 
If so how would it be interfaced/configured?
 
Thanks,
 
Clifford 
 
 
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From saeedahmad1981 at gmail.com  Mon Dec 15 07:25:43 2008
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 15 Dec 2008 16:25:43 +0100
Subject: [Freeswitch-users] Newbie Questions
Message-ID: <293ED6E2D87647248E526DFA0C08462C@SaeedLaptop>

Hi,

I am very new Freeswitch.

Till now I've some experience with Asterisk.

Can someone explain me the following things:

1. Can I connect my TDM switch to Freeswitch? My switch can speak Dss1/ss7,
SS7 is more preferable 
2. Is freeswitch similar to nextone SBC?
(http://www.nextpointnetworks.com/)
3. Does freeswtich support codec translation?

These are very basic questions at start. When I'll go deeper into it then
there could be more questions.

Kind Regards
Saeed



From jflowers at ezo.net  Mon Dec 15 07:45:38 2008
From: jflowers at ezo.net (jflowers)
Date: Mon, 15 Dec 2008 07:45:38 -0800 (PST)
Subject: [Freeswitch-users]  Speed Dial Emulation
Message-ID: <21016167.post@talk.nabble.com>


How do I emulate a speed dial setup.  That is, from extension 1003 I dial
just a 1 ( or 2, or 3 etc.) and nothing else and freeswitch dials a PSTN
number.  Is there software to do this?


-- 
View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21016167.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From anthony.minessale at gmail.com  Mon Dec 15 07:48:08 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Dec 2008 09:48:08 -0600
Subject: [Freeswitch-users] Interrupting read application with DTMF
In-Reply-To: <698401620812150127s4f96a7c4h3516da007f7399a0@mail.gmail.com>
References: <698401620812150127s4f96a7c4h3516da007f7399a0@mail.gmail.com>
Message-ID: <191c3a030812150748o60877e95yf071302c37d55b4@mail.gmail.com>

I think your solution is most likely superstition and that your real problem
is related
to your cell phone and the PSTN to SIP translation somewhere along the way.

I bet if you called in to the same extension with a SIP desk phone, that it
would work every
time no matter what format your file is.


On Mon, Dec 15, 2008 at 3:27 AM, Jan Kubr <jan.kubr at gmail.com> wrote:

> Hi,
> I have been having some troubles with the read application for quite a
> while which I haven't been able to solve yet.
> I have Freeswitch connected to a SIP gateway to accept calls from a
> landline-like number. For the incoming calls I have a simple testing
> dialplan:
>
> <action application="read" data="1 1
> en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
> <action application="phrase" data="spell,${res}"/>
>
> The behavior I have a problem with is that the read app should
> terminate when I press a digit and the execution should jump to the
> next action - meaning the playback of the file should be interrupted.
> The problem is that when I call the public number from my cell phone
> this works only about 50% of the time. In the other cases I need to
> wait for the wav file to be played (or press the digit two or three
> times). When using a SIP phone it always works.
>
> Today I tried to convert the wav file the read app plays to the GSM
> format and found out it fixed the problem! Now I can almost always
> interrupt the read app with DTMF from my cell phone. Doing the same
> from my SIP phone doesn't work well though when the file is GSM.
>
> Can someone explain me what is going on here and what is the right
> approach? I'm on revision 10751. I've tried to set a few configuration
> variables based on suggestions from this list, but it didn't make any
> difference.
>
> Thanks,
> Jan Kubr
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From wasim at convergence.pk  Mon Dec 15 07:49:19 2008
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 15 Dec 2008 20:49:19 +0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <293ED6E2D87647248E526DFA0C08462C@SaeedLaptop>
References: <293ED6E2D87647248E526DFA0C08462C@SaeedLaptop>
Message-ID: <b8ad2a5b0812150749t75e3c058u77f05fc5a91b5bbe@mail.gmail.com>

On Mon, Dec 15, 2008 at 8:25 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

Hi,


Salaam Saeed.


> I am very new Freeswitch.


Welcome.


> Till now I've some experience with Asterisk.


Be prepared to be amazed.

Can someone explain me the following things:
>
> 1. Can I connect my TDM switch to Freeswitch? My switch can speak Dss1/ss7,
> SS7 is more preferable


Currently, there is no open source ss7 implementation for FS. You can use
Sangoma's SMG with FS though.


> 2. Is freeswitch similar to nextone SBC?
> (http://www.nextpointnetworks.com/)


It can act as an SBC. See http://wiki.freeswitch.org/wiki/Specsheet
for more details.

3. Does freeswtich support codec translation?


Yes, it does for the supported codecs http://wiki.freeswitch.org/wiki/Codecs


> These are very basic questions at start. When I'll go deeper into it then
> there could be more questions.


Do read up at http://wiki.freeswitch.org/wiki/Main_Page

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as
you scope creep, so shall we reap ...
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From gmaruzz at celliax.org  Mon Dec 15 07:58:35 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 15 Dec 2008 16:58:35 +0100
Subject: [Freeswitch-users] Multiple mod_portaudio channels
In-Reply-To: <49467c03.030bca0a.2a43.ffff90cfSMTPIN_ADDED@mx.google.com>
References: <49467c03.030bca0a.2a43.ffff90cfSMTPIN_ADDED@mx.google.com>
Message-ID: <7b197bef0812150758r611e44c3re692902213891da6@mail.gmail.com>

You can try making the lynx appear to Operating System like 8 single
soundcards (mono 1in 1out).
You can do this on Linux via the /etc/asound.conf

Then for each soundcard you created, you can start one portaudio channel


Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Mon, Dec 15, 2008 at 11:56 AM, Clifford <freeswitch at ptmm.com> wrote:
> I need to set up a system that would allow multiple mod_portaudio channels
> to be configured.
>
> The calls would come in over SIP and any calls would be assigned a specific
> port on the multi-port sound card.
>
> I would probably set it up so specific extensions go to specific ports, for
> example 1001 would be port 1 and 1008 would be port 8 on the sound card.
>
> The documentation on the mod_portaudio module is very slim.  In fact I would
> say it is nearly undocumented.  Is it capable of handling multiple audio
> device ports (such as an 8-port LYNX sound card)?
>
> If so how would it be interfaced/configured?
>
> Thanks,
>
> Clifford
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jonas.gauffin at gmail.com  Mon Dec 15 08:01:01 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 15 Dec 2008 17:01:01 +0100
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
	<20081215110346.GA12681@jdc.jasonjgw.net>
	<432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>
Message-ID: <d021d3be0812150801g3d904fb5xe98b281f142a2d95@mail.gmail.com>

Yeah I know. That's what I'm trying to do, but with the domain name (
sip-corporate2.tele2.se) instead of the  ip.
I'm not sure that the gateway works without authentication and I'm wondering
if MANDATORY_IE_MISSING means that the gateway wants authentication or if it
means something else.
It's the "No Matching gateway found" message that is confusing, since I'm
not trying to use a registered gateway?

On Mon, Dec 15, 2008 at 4:05 PM, Brian West <brian at freeswitch.org> wrote:

> If you don't need auth you don't need a gateway.  sofia/profile/
> number at remoteip is all you should need.
>
> /b
>
> On Dec 15, 2008, at 5:03 AM, Jason White wrote:
>
> > On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
> >> I'm trying to bridge using a non-registered gateway. And I
> >> get MANDATORY_IE_MISSING back. Why is that?
> >
> > Does the gateway allow unauthenticated clients to make calls? If you
> > obtain a
> > SIP trace, you'll be able to see whether it's an authentication issue.
> >
> > As an aside, it would be an improvement to FreeSWITCH if Sofia
> > debugging could
> > be turned on and off within a running FreeSWITCH instance, including
> > SIP
> > traces, instead of the administrator's having to restart FreeSWITCH
> > with
> > environment variables exported, as is presently required according
> > to the
> > wiki.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From woof at nortel.com  Mon Dec 15 08:03:56 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 15 Dec 2008 11:03:56 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <87f2f3b90812131552p295eb1c4gee7d29f7a72624dc@mail.gmail.com>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
	<87f2f3b90812131552p295eb1c4gee7d29f7a72624dc@mail.gmail.com>
Message-ID: <op.ul7n8uh5ssieo1@woof.us.nortel.com>

Woof!

On Sat, 13 Dec 2008 18:52:59 -0500, Michael Collins <msc at freeswitch.org> wrote:

> All you'd have to do is modify the logfile.conf.xml file and pick a new path for your freeswitch.log file...

I agree.  I had discovered this option and considered it as a workaround.  Then I also found that mod_xml_rpc was also logging in log dir, and I haven't found a way to control that one--I haven't looked that hard, I must admit.  Weekends are a great way to forget everything you were doing the week before!


--Woof!


From rjcajax at gmail.com  Mon Dec 15 08:05:52 2008
From: rjcajax at gmail.com (Robert Clayton)
Date: Mon, 15 Dec 2008 11:05:52 -0500
Subject: [Freeswitch-users] Recording Pause
Message-ID: <d7e909fc0812150805k1f166ce4sdf4699c7a395f7e1@mail.gmail.com>

All,

I was thinking since there is no direct functionality for the person being
recorded to pause the recording could this be done indirectly.
For example if using the functionality to record only when voice is present
could the audio stream be preprocessed to evaluate the  dtmf where if a
specific key was pressed the audio stream would no longer be passed or
passed as blank to the recording functionality so an implicit pause would be
created. And when the person recording wished to continue pressing a key
would signal FS to pass the audio again therefore reestablishing recording
to the same file?

Bob


From chavpaskov at shaw.ca  Mon Dec 15 08:15:38 2008
From: chavpaskov at shaw.ca (Chav Paskov)
Date: Mon, 15 Dec 2008 08:15:38 -0800
Subject: [Freeswitch-users] how to handle  returned sip 302 dialplan
In-Reply-To: <49445591.90505@shaw.ca>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
	<49445591.90505@shaw.ca>
Message-ID: <494682AA.9090200@shaw.ca>

Chav Paskov wrote:
> Brian West wrote:
>   
>> Chav,
>> Once the 302 is received by FreeSWITCH it will follow it to the 
>> contact listed in the 302.  What else are you needing to do?
>>
>> /b
>>
>> On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
>>
>>     
>>> *User-Agent: eXosip/3.1.0^M
>>> Content-Length:  
>>>
>>>
>>> my question is:
>>>
>>> Is it possible to send the call to z.z.z.z , receive the SIP 302 , 
>>> process the data in Contact field  and redirect to the new destination 
>>> contained in *Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
>>> *without closing the session.
>>> i red something about   <action application="set" 
>>> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
>>>
>>> Any ideas on this matter  will be highly appreciated.
>>> Best Regards
>>> Chav
>>>
>>>       
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
> Thanks Brian,
>
> probably i should have explained it in more details.
> this whole thing started as an attempt to  implement lata ocn /local 
> number portability/ instead of  pure per destination routing.
> At the moment i have a access to  a service provider  who does 
> "dipping"  and returns  the  LATA OCN  data associated with  any dialed 
> destination number. it is returned as Contact:  and Content-length:  
> fields in 302 message.
>
> in other words:
>
> 1. i'm sending to this provider let say - 2025556666  as a destination 
> number.
> 2. they do the dipping  and will return to me  either the new dest #  if 
> 2025556666 has been ported or if it was not
> in content-length field they'll send  lata, ocn  and state and 10 digits 
> number.
> 3. once received i have to  compare the received lata , ocn and state  
> date with  a compiled  rate deck / blended from 5 different vendors/  
> and pick the lowest rate - effectively building  LCR based on LATA OCN 
> STATE info.
>
> Hope this will help to clear the picture.
> Regards
> Chav
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From sicfslist at gmail.com  Mon Dec 15 08:40:13 2008
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Mon, 15 Dec 2008 10:40:13 -0600
Subject: [Freeswitch-users] how to handle returned sip 302 dialplan
In-Reply-To: <49445591.90505@shaw.ca>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
	<49445591.90505@shaw.ca>
Message-ID: <35b355e90812150840k362096cah292647c92b8681a0@mail.gmail.com>

Chav,
We recently / are still going through the same process (in order to route on
LRN) vs NPANXX or LATA based routing.  Here was the best way that we came up
with to do it:
  -- we use xml_curl exclusively for routing decisions
  -- so in the cgi script that xml_curl hits one of the things it can (and
does based on certain parameters) is fire off a url to another LNP server
that we built
  -- the LNP server actually does the dip (either from a cache) and returns
the info

We felt this was much better for a few reasons:
  -- caching the LNP data for a 24 hour period would save us in excess of
$100k a year
  -- having a specialized mechanism to do this was much easier to implement
for the cgi process than supporting 302 redirects directly on the FS boxes
was much easier (which just wasn't possible with the cgi mechanism)
  -- every LNP provider returns 302's slightly different ... so we didn't
want to have to reinvent the wheel on the FS machines if we ever wanted to
add redundancy or switch providers

Guess it all depends on your config ... but this was the easiest and most
cost-effective means for us to implement.


On Sat, Dec 13, 2008 at 6:38 PM, Chav Paskov <chavpaskov at shaw.ca> wrote:

> Brian West wrote:
> > Chav,
> > Once the 302 is received by FreeSWITCH it will follow it to the
> > contact listed in the 302.  What else are you needing to do?
> >
> > /b
> >
> > On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
> >
> >> *User-Agent: eXosip/3.1.0^M
> >> Content-Length:
> >>
> >>
> >> my question is:
> >>
> >> Is it possible to send the call to z.z.z.z , receive the SIP 302 ,
> >> process the data in Contact field  and redirect to the new destination
> >> contained in *Contact: <sip:2145551234 at 64.112.188.84<sip%3A2145551234 at 64.112.188.84>
> >;npdi^M
> >> *without closing the session.
> >> i red something about   <action application="set"
> >> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
> >>
> >> Any ideas on this matter  will be highly appreciated.
> >> Best Regards
> >> Chav
> >>
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> Thanks Brian,
>
> probably i should have explained it in more details.
> this whole thing started as an attempt to  implement lata ocn /local
> number portability/ instead of  pure per destination routing.
> At the moment i have a access to  a service provider  who does
> "dipping"  and returns  the  LATA OCN  data associated with  any dialed
> destination number. it is returned as Contact:  and Content-length:
> fields in 302 message.
>
> in other words:
>
> 1. i'm sending to this provider let say - 2025556666  as a destination
> number.
> 2. they do the dipping  and will return to me  either the new dest #  if
> 2025556666 has been ported or if it was not
> in content-length field they'll send  lata, ocn  and state and 10 digits
> number.
> 3. once received i have to  compare the received lata , ocn and state
> date with  a compiled  rate deck / blended from 5 different vendors/
> and pick the lowest rate - effectively building  LCR based on LATA OCN
> STATE info.
>
> Hope this will help to clear the picture.
> Regards
> Chav
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saeedahmad1981 at gmail.com  Mon Dec 15 08:42:25 2008
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 15 Dec 2008 17:42:25 +0100
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <b8ad2a5b0812150749t75e3c058u77f05fc5a91b5bbe@mail.gmail.com>
References: <293ED6E2D87647248E526DFA0C08462C@SaeedLaptop>
	<b8ad2a5b0812150749t75e3c058u77f05fc5a91b5bbe@mail.gmail.com>
Message-ID: <FB4BFC95627B448DA2627A2321DE6C9C@SaeedLaptop>

Wsalam Waseem sb.

Good to see you on mailing list :-)

 

So i hope you already know why I am trying to move to FS, I am still trying
to do similar thing, blind transfer of zap call with number changed (22)
release cause with new number, since I'll do it with freeswitch do you think
its possible now?

- Saeed

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wasim
Baig
Sent: Monday, December 15, 2008 4:49 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie Questions

 

On Mon, Dec 15, 2008 at 8:25 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>
wrote:

Hi,


Salaam Saeed.
 

I am very new Freeswitch.


Welcome.
 

Till now I've some experience with Asterisk.


Be prepared to be amazed. 

Can someone explain me the following things:

1. Can I connect my TDM switch to Freeswitch? My switch can speak Dss1/ss7,
SS7 is more preferable


Currently, there is no open source ss7 implementation for FS. You can use
Sangoma's SMG with FS though.
 

2. Is freeswitch similar to nextone SBC?
(http://www.nextpointnetworks.com/)


It can act as an SBC. See http://wiki.freeswitch.org/wiki/Specsheet
for more details.

3. Does freeswtich support codec translation?


Yes, it does for the supported codecs http://wiki.freeswitch.org/wiki/Codecs
 

These are very basic questions at start. When I'll go deeper into it then
there could be more questions.


Do read up at http://wiki.freeswitch.org/wiki/Main_Page 


-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as
you scope creep, so shall we reap ... 

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From chavpaskov at shaw.ca  Mon Dec 15 08:45:22 2008
From: chavpaskov at shaw.ca (Chav Paskov)
Date: Mon, 15 Dec 2008 08:45:22 -0800
Subject: [Freeswitch-users] how to handle returned sip 302 dialplan
In-Reply-To: <35b355e90812150840k362096cah292647c92b8681a0@mail.gmail.com>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
	<49445591.90505@shaw.ca>
	<35b355e90812150840k362096cah292647c92b8681a0@mail.gmail.com>
Message-ID: <494689A2.7080303@shaw.ca>

Shelby Ramsey wrote:
> Chav,
>
> We recently / are still going through the same process (in order to 
> route on LRN) vs NPANXX or LATA based routing.  Here was the best way 
> that we came up with to do it:
>   -- we use xml_curl exclusively for routing decisions
>   -- so in the cgi script that xml_curl hits one of the things it can 
> (and does based on certain parameters) is fire off a url to another 
> LNP server that we built
>   -- the LNP server actually does the dip (either from a cache) and 
> returns the info
>
> We felt this was much better for a few reasons:
>   -- caching the LNP data for a 24 hour period would save us in excess 
> of $100k a year
>   -- having a specialized mechanism to do this was much easier to 
> implement for the cgi process than supporting 302 redirects directly 
> on the FS boxes was much easier (which just wasn't possible with the 
> cgi mechanism)
>   -- every LNP provider returns 302's slightly different ... so we 
> didn't want to have to reinvent the wheel on the FS machines if we 
> ever wanted to add redundancy or switch providers
>
> Guess it all depends on your config ... but this was the easiest and 
> most cost-effective means for us to implement.
>
>
> On Sat, Dec 13, 2008 at 6:38 PM, Chav Paskov <chavpaskov at shaw.ca 
> <mailto:chavpaskov at shaw.ca>> wrote:
>
>     Brian West wrote:
>     > Chav,
>     > Once the 302 is received by FreeSWITCH it will follow it to the
>     > contact listed in the 302.  What else are you needing to do?
>     >
>     > /b
>     >
>     > On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
>     >
>     >> *User-Agent: eXosip/3.1.0^M
>     >> Content-Length:
>     >>
>     >>
>     >> my question is:
>     >>
>     >> Is it possible to send the call to z.z.z.z , receive the SIP 302 ,
>     >> process the data in Contact field  and redirect to the new
>     destination
>     >> contained in *Contact: <sip:2145551234 at 64.112.188.84
>     <mailto:sip%3A2145551234 at 64.112.188.84>>;npdi^M
>     >> *without closing the session.
>     >> i red something about   <action application="set"
>     >> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
>     >>
>     >> Any ideas on this matter  will be highly appreciated.
>     >> Best Regards
>     >> Chav
>     >>
>     >
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > Freeswitch-users mailing list
>     > Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     Thanks Brian,
>
>     probably i should have explained it in more details.
>     this whole thing started as an attempt to  implement lata ocn /local
>     number portability/ instead of  pure per destination routing.
>     At the moment i have a access to  a service provider  who does
>     "dipping"  and returns  the  LATA OCN  data associated with  any
>     dialed
>     destination number. it is returned as Contact:  and Content-length:
>     fields in 302 message.
>
>     in other words:
>
>     1. i'm sending to this provider let say - 2025556666  as a destination
>     number.
>     2. they do the dipping  and will return to me  either the new dest
>     #  if
>     2025556666 has been ported or if it was not
>     in content-length field they'll send  lata, ocn  and state and 10
>     digits
>     number.
>     3. once received i have to  compare the received lata , ocn and state
>     date with  a compiled  rate deck / blended from 5 different vendors/
>     and pick the lowest rate - effectively building  LCR based on LATA OCN
>     STATE info.
>
>     Hope this will help to clear the picture.
>     Regards
>     Chav
>
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
thanks for the prompt response.
Can you pls  give me an example how to access the info  contained in  
Contact:  and content-legth: fields  if you can.
I was thinking  in going the exactly same direction  in terms of 
building  xml_curl  dialplan  but i'm lacking knowledge
on how to access variables.
Regards
Chav


From intralanman at freeswitch.org  Mon Dec 15 08:53:49 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Mon, 15 Dec 2008 11:53:49 -0500
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <d021d3be0812150801g3d904fb5xe98b281f142a2d95@mail.gmail.com>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>	<20081215110346.GA12681@jdc.jasonjgw.net>	<432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>
	<d021d3be0812150801g3d904fb5xe98b281f142a2d95@mail.gmail.com>
Message-ID: <49468B9D.5010903@freeswitch.org>

posting relevant pieces of your dialplan and sofia configs would 
probably help a bit.

-Ray


Jonas Gauffin wrote:
> Yeah I know. That's what I'm trying to do, but with the domain name 
> (sip-corporate2.tele2.se <http://sip-corporate2.tele2.se>) instead of 
> the  ip.
>
> I'm not sure that the gateway works without authentication and I'm 
> wondering if MANDATORY_IE_MISSING means that the gateway wants 
> authentication or if it means something else.
> It's the "No Matching gateway found" message that is confusing, since 
> I'm not trying to use a registered gateway?
>
> On Mon, Dec 15, 2008 at 4:05 PM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     If you don't need auth you don't need a gateway.  sofia/profile/
>     number at remoteip is all you should need.
>
>     /b
>
>     On Dec 15, 2008, at 5:03 AM, Jason White wrote:
>
>     > On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
>     >> I'm trying to bridge using a non-registered gateway. And I
>     >> get MANDATORY_IE_MISSING back. Why is that?
>     >
>     > Does the gateway allow unauthenticated clients to make calls? If you
>     > obtain a
>     > SIP trace, you'll be able to see whether it's an authentication
>     issue.
>     >
>     > As an aside, it would be an improvement to FreeSWITCH if Sofia
>     > debugging could
>     > be turned on and off within a running FreeSWITCH instance, including
>     > SIP
>     > traces, instead of the administrator's having to restart FreeSWITCH
>     > with
>     > environment variables exported, as is presently required according
>     > to the
>     > wiki.
>     >
>     >
>     > _______________________________________________
>     > Freeswitch-users mailing list
>     > Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From brian at freeswitch.org  Mon Dec 15 08:55:10 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 10:55:10 -0600
Subject: [Freeswitch-users] how to handle returned sip 302 dialplan
In-Reply-To: <494689A2.7080303@shaw.ca>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
	<49445591.90505@shaw.ca>
	<35b355e90812150840k362096cah292647c92b8681a0@mail.gmail.com>
	<494689A2.7080303@shaw.ca>
Message-ID: <97AF9163-4EE6-43D6-A232-A0534519F723@freeswitch.org>

I fear that you won't be able to get at any information in a 302  
issued to FreeSWITCH as those are on auto pilot.

You can try toying with the info application to see if it gets at the  
info you need.

/b

On Dec 15, 2008, at 10:45 AM, Chav Paskov wrote:

> thanks for the prompt response.
> Can you pls  give me an example how to access the info  contained in
> Contact:  and content-legth: fields  if you can.
> I was thinking  in going the exactly same direction  in terms of
> building  xml_curl  dialplan  but i'm lacking knowledge
> on how to access variables.
> Regards
> Chav



From brian at freeswitch.org  Mon Dec 15 08:56:53 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 10:56:53 -0600
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <op.ulwl5dbessieo1@woof.us.nortel.com>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
Message-ID: <F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>

I can't figure out why the log file would need to be in the db folder...

/b

On Dec 9, 2008, at 10:45 AM, Andy Spitzer wrote:

> Woof!
>
> It appears that FreeSWITCH writes
>
>   freeswitch.history
>   freeswitch.log
>   freeswitch.pid
>   freeswitch.xml.fsxml
>
> to the -log directory.
>
> Is there a way to put the files other than freeswitch.log into the - 
> db directory instead?
>
> In my environment we archive and rotate everything in the log  
> directory (which includes logs beside FreeSWITCH's), and these other  
> FreeSWITCH files are getting rotated.  Yeah, I can explicitly  
> exclude them, but to me it seems those really belong in the -db  
> directory anyway, as they are inherently data needed for the current  
> executable of FreeSWITCH, and not logs.
>
> --Woof!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From stevecrozz at gmail.com  Mon Dec 15 09:25:58 2008
From: stevecrozz at gmail.com (Stephen Crosby)
Date: Mon, 15 Dec 2008 09:25:58 -0800
Subject: [Freeswitch-users] running custom script with bind_meta_app
In-Reply-To: <910AFB7A-9CDE-49AE-A72D-7942F8F79DA3@freeswitch.org>
References: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>
	<910AFB7A-9CDE-49AE-A72D-7942F8F79DA3@freeswitch.org>
Message-ID: <11990ade0812150925i33e9cf6ex6a1fe53c423fae9b@mail.gmail.com>

I just want to listen for some DTMF sequence while in a conference.
The conference host should be able to enter the sequence at any time
(and any number of times) to run a custom script. I've already written
one in javascript, but I can rewrite it in another language if it's
easier.

On Mon, Dec 15, 2008 at 7:09 AM, Brian West <brian at freeswitch.org> wrote:
> What are you wanting to accomplish first?
> /b
> On Dec 15, 2008, at 1:32 AM, Stephen Crosby wrote:
>
> I'm Stephen Crosby, and I've just started working with freeswitch.
> It's been great so far.
>
> I want to run a custom script inside a conference when a DTMF sequence
> is entered. I found bind_meta_app and thought it would be perfect, but
> I can't seem to get it to work. When I dial-in and press *8, I get no
> debugging output at all. When I press another sequence like *9 for
> instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf()
> sofia/external/5593495805 at sip.gafachi.com Ignoring meta digit '9' not
> mapped. The script I wrote has been tested with "jsrun script.js" from
> the command line and it does work. I've got the debugging level all
> the way up and there's just not much for me to go on. Any help would
> be greatly appreciated.
>
> <extension name="gafachi-DID1">
>  <condition field="destination_number" expression="^xxxxxxxxxxx$">
>    <action application="answer"/>
>    <action application="bind_meta_app" data="8 ab s
> javascript::$${base_dir}/scripts/script.js"/>
>    <action application="conference" data="3000 at default"/>
>  </condition>
> </extension>
>
> --Stephen
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From woof at nortel.com  Mon Dec 15 09:35:24 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 15 Dec 2008 12:35:24 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
	<F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>
Message-ID: <op.ul7shal3ssieo1@woof.us.nortel.com>

Woof!

On Mon, 15 Dec 2008 11:56:53 -0500, Brian West <brian at freeswitch.org> wrote:

> I can't figure out why the log file would need to be in the db folder...

I think you misunderstand.

It's these files:
   freeswitch.history
   freeswitch.pid
   freeswitch.xml.fsxml

That I feel would be better off in the db folder.  They are not logs, and should not be rotated.

--Woof!


From intralanman at freeswitch.org  Mon Dec 15 10:16:32 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Mon, 15 Dec 2008 13:16:32 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <op.ul7shal3ssieo1@woof.us.nortel.com>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>	<F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>
	<op.ul7shal3ssieo1@woof.us.nortel.com>
Message-ID: <49469F00.7020203@freeswitch.org>


> I think you misunderstand.
>
> It's these files:
>    freeswitch.history
>    freeswitch.pid
>    freeswitch.xml.fsxml
>
> That I feel would be better off in the db folder.  They are not logs, and should not be rotated.
>
>   


if freeswitch.history isn't a log, what is it? seems to me taht it's a 
log of what commands you've run recently... it's definitely NOT a 
database.... neither is the pid file. while the pid file isn't a db, 
it's also not really a log... but i don't know that i'd agree with 
making a "run" directory just to house the pid.

-Ray


From kkielhofner at star2star.com  Mon Dec 15 10:35:18 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Mon, 15 Dec 2008 13:35:18 -0500
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215080105.GA11352@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
Message-ID: <2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>

On Mon, Dec 15, 2008 at 3:01 AM, Jason White <jason at jasonjgw.net> wrote:
> Turning on all of the Sofia debugging options reveals the following.
>
> FS sends out the invite (UDP, 1245 bytes).
>
> Then we get:
>
> nta: timer shortened to 500 ms
> tport_wakeup_pri(0x2490700): events ERR
> tport_udp_error: icmp(6) message was truncated (at 832)
> tport_udp_error: Message too long (90) [icmp6 type=2 code=0 info=00000500]
>        reported by [2001:470:0:5d::2]:65535
> tport_release(0x2490700): 0x7fe244098c10 by 0x25022a0 with (nil)
> nta: INVITE (108544297): Message too long (90) with udp/[xxxxx - remote host's
> IPv6 address]:5060
> nua(0x7fe24409af90): event r_invite 503 Service Unavailable
> nua(0x7fe24409af90): call state changed: calling -> init
> nua: nua_application_event: entering
> nua(0x7fe24409af90): event i_state 503 Service Unavailable
> nua(0x7fe24409af90): event i_terminated 503 Service Unavailable
> nua: nua_handle_magic: entering
> nua(0x7fe24409af90): removing session usage
> nua: nua_application_event: entering
> nta_leg_destroy(0x7fe2440822f0)
> 2008-12-15 18:14:27 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/external-ipv6/nnnn at remote-host.domain entering state [terminated]
>
> The IPv6 address that sent the "message too long" error is presumably that of
> a router between my FreeSWITCH box and my friend's FreeSWITCH box.
>
> Is there a way around this?

What happens if you use TCP transport instead of UDP?  You're probably
running into the infamous SIP UDP fragmentation (there isn't any)
problem.


-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From woof at nortel.com  Mon Dec 15 10:37:15 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 15 Dec 2008 13:37:15 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <49469F00.7020203@freeswitch.org>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
	<F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>
	<op.ul7shal3ssieo1@woof.us.nortel.com>
	<49469F00.7020203@freeswitch.org>
Message-ID: <op.ul7vcd14ssieo1@woof.us.nortel.com>

Woof!

On Mon, 15 Dec 2008 13:16:32 -0500, Raymond Chandler <intralanman at freeswitch.org> wrote:

> if freeswitch.history isn't a log, what is it? seems to me taht it's a
> log of what commands you've run recently... it's definitely NOT a
> database....

Actually, I the readline/history library uses it to determine the command line history (http://tiswww.case.edu/php/chet/readline/history.html#SEC15).  So it IS a database.  It may also be seen as a log of commands, as it happens to be a nice ASCII file, but that's an intended side effect of the way the library writes it.


--Woof!


From brian at freeswitch.org  Mon Dec 15 10:43:08 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 12:43:08 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>
Message-ID: <4D744169-0A93-44C7-AEDF-89C78F279AE2@freeswitch.org>

Well its important to know if he's on 1.0.1 or SVN trunk.  In 1.0.1  
the retry via TCP was disabled on ipv6 and that has now been  
corrected.  If it gets the message too big error it'll turn around and  
requeue the invite on TCP without any user interaction.

/b

On Dec 15, 2008, at 12:35 PM, Kristian Kielhofner wrote:

>
> What happens if you use TCP transport instead of UDP?  You're probably
> running into the infamous SIP UDP fragmentation (there isn't any)
> problem.



From brian at freeswitch.org  Mon Dec 15 11:46:07 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 13:46:07 -0600
Subject: [Freeswitch-users] ./configure fails on 10.4.11
In-Reply-To: <B9DF8866-2120-4E5C-8FFB-352A11A71438@stillnewt.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
	<830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
	<52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>
	<1848C35A-040F-4365-AA17-E43C9B32E11D@freeswitch.org>
	<B9DF8866-2120-4E5C-8FFB-352A11A71438@stillnewt.org>
Message-ID: <A9592378-9B8D-41D1-82D1-1ACE1CF3E165@freeswitch.org>

Anyway you can get one of us onto that OS X machine?  We haven't been  
able to reproduce this and without access to a machine its taking  
place on we can't fix it.

/b

On Dec 13, 2008, at 6:23 PM, martin joseph wrote:

>
> On Dec 13, 2008, at 2:07 PM, Brian West wrote:
>
>> Marty,
>> 	Can you point out where its failing?
> ./configure fails as follows (just as in the JIRA)
>
> checking for a BSD-compatible install... /usr/bin/install -c
> ./configure: line 4112: syntax error near unexpected token
> `build_libtool_libs,'
> ./configure: line 4112: `    _LT_DECL(build_libtool_libs,
> enable_shared, 0,'
> configure: error: ./configure.gnu failed for libs/openmrcp
>
>> Nobody has been able to
>> reproduce the issue that was reported on jira.  Even Anthony can't  
>> and
>> he's on 10.4.
> Huh,  I would love to figure this out, as it seems certain to be
> somehow specific to my install.
>>  I'm on 10.5 and I don't have any issues either.  So if
>> you can pin point the exact place where it fails we can look at it
>> closer.
> I would love that.
>
> Thanks again for your excellent software and help.
> Marty
>
>>
>> /b
>>
>> On Dec 13, 2008, at 3:59 PM, Martin Joseph wrote:
>>
>>> Yes, I think it looks good too.  But make fails with the Jira issue
>>> that has been going on for ages.
>>>
>>> I have never used fink or darwinports or any other such methods on
>>> this box so that's out.
>>>
>>> I definitely do have the Apple devtools for 10.4 installed.
>>>
>>> I have no problem making the 1.01 FS from the tarball, but as it
>>> seems you are telling everyone to upgrade to the SVN trunk, I would
>>> love to do that also.  However, I am frustrated by my inability to
>>> get that going, as well as a severe lack of time.
>>>
>>> Thanks for any help or ideas,
>>> Marty
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kkielhofner at star2star.com  Mon Dec 15 12:06:39 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Mon, 15 Dec 2008 15:06:39 -0500
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <4D744169-0A93-44C7-AEDF-89C78F279AE2@freeswitch.org>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>
	<4D744169-0A93-44C7-AEDF-89C78F279AE2@freeswitch.org>
Message-ID: <2d9149cd0812151206y8df6d3cn8cb998e914698019@mail.gmail.com>

On Mon, Dec 15, 2008 at 1:43 PM, Brian West <brian at freeswitch.org> wrote:
> Well its important to know if he's on 1.0.1 or SVN trunk.  In 1.0.1
> the retry via TCP was disabled on ipv6 and that has now been
> corrected.  If it gets the message too big error it'll turn around and
> requeue the invite on TCP without any user interaction.
>
> /b
>

...which is exactly what the RFC says to do (IPv6 or IPv4).

Bravo!

I thought (for the user) it might make sense to just force TCP.  It
might make sense to set TCP explicitly (less back and forth).
However, keep in mind the higher overhead of TCP in the first place.
I guess it would depend on what's in the body (obviously).  Messages
with smaller bodies (SDPs, etc) might eek through with UDP while those
with larger bodies may have to requeue for TCP.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From brian at freeswitch.org  Mon Dec 15 12:14:50 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 14:14:50 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <2d9149cd0812151206y8df6d3cn8cb998e914698019@mail.gmail.com>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>
	<4D744169-0A93-44C7-AEDF-89C78F279AE2@freeswitch.org>
	<2d9149cd0812151206y8df6d3cn8cb998e914698019@mail.gmail.com>
Message-ID: <114ADAD5-3964-431B-8E47-D8B2FD8E7393@freeswitch.org>

Pretty much everything in ipv6 will be over the MTU...  on ipv4 we  
disabled this because when you talk to things that don't speak TCP you  
have a packet over the MTU you try TCP it times out in 30 seconds then  
you send it anyway via UDP.  So why have a 30 seconds timeout when you  
don't need too.

/b

On Dec 15, 2008, at 2:06 PM, Kristian Kielhofner wrote:

> I thought (for the user) it might make sense to just force TCP.  It
> might make sense to set TCP explicitly (less back and forth).
> However, keep in mind the higher overhead of TCP in the first place.
> I guess it would depend on what's in the body (obviously).  Messages
> with smaller bodies (SDPs, etc) might eek through with UDP while those
> with larger bodies may have to requeue for TCP.



From jason at jasonjgw.net  Mon Dec 15 14:54:51 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Dec 2008 09:54:51 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
Message-ID: <20081215225451.GA5592@jdc.jasonjgw.net>

On Mon, Dec 15, 2008 at 09:08:56AM -0600, Brian West wrote:
>  From the past emails and the data you have provided so far it makes  
> me believe you're not on SVN trunk.  

I'm on trunk as of December 12, revision 10725. If it was fixed since then, I
can easily recompile.

The other party is on a recent revision from trunk and also runs into the same
problems in trying to call me via IPv6.



From brian at freeswitch.org  Mon Dec 15 15:09:21 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 17:09:21 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215225451.GA5592@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
	<20081215225451.GA5592@jdc.jasonjgw.net>
Message-ID: <0DEA56F4-9C15-47FA-8746-924CD6EF4109@freeswitch.org>

Well I just tested this again with the latest svn trunk

d77374e2-a381-4eb1-8771-5ccdae2155fd,2008-12-15 17:05:51,1229382351,sofia/internal/1000 at bkw.org 
,CS_EXECUTE,Brian West,1000,99.157.44.200,9888,bridge,sofia/internal- 
ipv6/888@[2001:470:7:ea::2],XML,default,G722,16000,G722,16000
0ed0b7ee-5de5-44a3-9f96-f6bbc5a7e0ba,2008-12-15  
17:05:51,1229382351,sofia/internal-ipv6/888@[2001:470:7:ea:: 
2],CS_EXCHANGE_MEDIA,Brian West,1000,99.157.44.200,888@[2001:470:7:ea:: 
2],,,XML,default,G722,16000,G722,16000

   1.447392 2001:470:1f0f:142:21a:92ff:fe3f:6a0f -> 2001:470:7:ea::2  
RTP PT=ITU-T G.722, SSRC=0x89B50B10, Seq=64494, Time=2387361962
   1.460530 2001:470:7:ea::2 -> 2001:470:1f0f:142:21a:92ff:fe3f:6a0f  
RTP PT=ITU-T G.722, SSRC=0xEB44C067, Seq=37131, Time=1328480

Seems to work fine for me... care to let me at your machine to take a  
look?

Works great did an ipv4 call to my FS box and ipv6 out to the  
conference box.


Try this out

sofia/internal-ipv6/888@[2001:470:7:ea::2]

(correct the profile for your needs)

/b

On Dec 15, 2008, at 4:54 PM, Jason White wrote:

> On Mon, Dec 15, 2008 at 09:08:56AM -0600, Brian West wrote:
>> From the past emails and the data you have provided so far it makes
>> me believe you're not on SVN trunk.
>
> I'm on trunk as of December 12, revision 10725. If it was fixed  
> since then, I
> can easily recompile.
>
> The other party is on a recent revision from trunk and also runs  
> into the same
> problems in trying to call me via IPv6.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason at jasonjgw.net  Mon Dec 15 15:29:56 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Dec 2008 10:29:56 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <0DEA56F4-9C15-47FA-8746-924CD6EF4109@freeswitch.org>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
	<20081215225451.GA5592@jdc.jasonjgw.net>
	<0DEA56F4-9C15-47FA-8746-924CD6EF4109@freeswitch.org>
Message-ID: <20081215232956.GA6139@jdc.jasonjgw.net>

On Mon, Dec 15, 2008 at 05:09:21PM -0600, Brian West wrote:
> sofia/internal-ipv6/888@[2001:470:7:ea::2]
> 
> (correct the profile for your needs)

That worked straight away.

At least it shows that there's nothing wrong with the IPv6 connectivity as
such. I'll try to isolate what is different between this and the call that I
was attempting yesterday.



From brian at freeswitch.org  Mon Dec 15 15:36:51 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 17:36:51 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215232956.GA6139@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
	<20081215225451.GA5592@jdc.jasonjgw.net>
	<0DEA56F4-9C15-47FA-8746-924CD6EF4109@freeswitch.org>
	<20081215232956.GA6139@jdc.jasonjgw.net>
Message-ID: <2DC959F8-9A78-4D1D-8C6C-792A401689CE@freeswitch.org>

btw thats on an HE tunnel too ;)

/b

On Dec 15, 2008, at 5:29 PM, Jason White wrote:

> That worked straight away.
>
> At least it shows that there's nothing wrong with the IPv6  
> connectivity as
> such. I'll try to isolate what is different between this and the  
> call that I
> was attempting yesterday.



From c_cav_01 at yahoo.com  Mon Dec 15 21:21:35 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Mon, 15 Dec 2008 21:21:35 -0800 (PST)
Subject: [Freeswitch-users]  Building a web config/billing gui
Message-ID: <21027515.post@talk.nabble.com>


For anyone interested, I'm in the process of building a web based
config/billing gui.  I could use some help though.  Anyone who has php
experience and knows the xml_curl interface pretty well and some spare time
to do some development would be a useful partner.

Also, if there's a resource online that defines all the params/variables and
subobjects for each of the fs object types, like directories, dialplans etc,
it would help me fill my parm database a lot faster.

For anyone interested in monitoring the development, it's at
www.sparkz.tv/smfs  login is demo:demo.  Feedback on features and
development direction is invited.  I want to get this done quickly.  I think
FS is awesome but no config/billing interface is going to stand in the way
of it's adoption so I'm burning the midnight oil to get this done.
-- 
View this message in context: http://www.nabble.com/Building-a-web-config-billing-gui-tp21027515p21027515.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From darren at aleph-com.net  Mon Dec 15 21:33:24 2008
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 15 Dec 2008 22:33:24 -0700
Subject: [Freeswitch-users] Building a web config/billing gui
In-Reply-To: <21027515.post@talk.nabble.com>
References: <21027515.post@talk.nabble.com>
Message-ID: <49473DA4.7090501@aleph-com.net>

Are you interested in joining an existing project?  We've been working 
on the ASTPP freeswitch port for a number of months.   The billing is 
all in place but  help with the same stuff you're looking for would help 
to get things fleshed out a little more. 

Darren Wiebe
darren at aleph-com.net

ccav wrote:
> For anyone interested, I'm in the process of building a web based
> config/billing gui.  I could use some help though.  Anyone who has php
> experience and knows the xml_curl interface pretty well and some spare time
> to do some development would be a useful partner.
>
> Also, if there's a resource online that defines all the params/variables and
> subobjects for each of the fs object types, like directories, dialplans etc,
> it would help me fill my parm database a lot faster.
>
> For anyone interested in monitoring the development, it's at
> www.sparkz.tv/smfs  login is demo:demo.  Feedback on features and
> development direction is invited.  I want to get this done quickly.  I think
> FS is awesome but no config/billing interface is going to stand in the way
> of it's adoption so I'm burning the midnight oil to get this done.
>   



From msc at freeswitch.org  Mon Dec 15 21:37:01 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 15 Dec 2008 21:37:01 -0800
Subject: [Freeswitch-users] Building a web config/billing gui
In-Reply-To: <21027515.post@talk.nabble.com>
References: <21027515.post@talk.nabble.com>
Message-ID: <3C01F130-D55F-4A1A-888C-98C556363AF6@freeswitch.org>

Ccav,

Thanks for helping out with the project! If you haven't already joined  
us on the irc channel please do so: #freeswitch on irc.freenode.net.  
Another channel you might be interested in is #tcapi. There is a group  
working on a general purpose GUI for FreeSWITCH at tcapi.org. They've  
probably faced some of the same challenges that you have.

We will be glad to help you however we possibly can.

-MC

Sent from my iPhone

On Dec 15, 2008, at 9:21 PM, ccav <c_cav_01 at yahoo.com> wrote:

>
> For anyone interested, I'm in the process of building a web based
> config/billing gui.  I could use some help though.  Anyone who has php
> experience and knows the xml_curl interface pretty well and some  
> spare time
> to do some development would be a useful partner.
>
> Also, if there's a resource online that defines all the params/ 
> variables and
> subobjects for each of the fs object types, like directories,  
> dialplans etc,
> it would help me fill my parm database a lot faster.
>
> For anyone interested in monitoring the development, it's at
> www.sparkz.tv/smfs  login is demo:demo.  Feedback on features and
> development direction is invited.  I want to get this done quickly.   
> I think
> FS is awesome but no config/billing interface is going to stand in  
> the way
> of it's adoption so I'm burning the midnight oil to get this done.
> -- 
> View this message in context: http://www.nabble.com/Building-a-web-config-billing-gui-tp21027515p21027515.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From Prometheus001 at gmx.net  Mon Dec 15 23:59:26 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 16 Dec 2008 08:59:26 +0100
Subject: [Freeswitch-users] mod_shout and mp3 formats
Message-ID: <49475FDE.7080108@gmx.net>

I try to play mp3 I generated through Cepstral TTs and which I encoded
via lame.
However they won't play, so my question is: Which mp3 formats are supported?

I generate the wav files by the following
/opt/swift/bin/swift -n Katrin -p
audio/channels=1,cst/f0_shift=.8,speech/rate=120,audio/sampling-rate=8000,audio/deadair=2
-o $wavefile $text

Then I convert to mp3 by the following variations:
lame 46.wav 46.mp3
lame -s 32 46.wav 46.mp3
lame --preset 128 46.wav 46.mp3
lame --resample 44.1 --preset 128 46.wav 46.mp3
lame --resample 32 --preset 128 46.wav 46.mp3
lame --resample 44.1 46.wav 46.mp3
lame --resample 44.1 -m s --preset 128 46.wav 46.mp3
lame --resample 44.1 -m s  46.wav 46.mp3
lame --resample 44.1 -m s -b 128 46.wav 46.mp3
lame --resample 44.1 -m s -B 24 46.wav 46.mp3
lame --preset voice -v -B 64 -a 46.wav 46.mp3

None of them worked with the playback application
(shout://localhost/tts/46.mp3). The sound files had a length of between
2 and 5 sec. 2 Times during various tries they played at least
partially. But at the next try they didn't play again. However I have a
prerecorded sound file (44.1KHz, 128 kBits stereo music) which always
plays well.
The console shows me that all files are successfully played and I get a
channel_ececute and a channel_ececute_complete after some seconds during
event_socket. But I don't hear any sound.

All above samples however played well with Totem on Ubuntu.

The wiki tells me that almost any mp3 format should play. What am I
doing wrong here?

Another question: Should normal wav files play as well? Also with wav I
cannot hear any sound.

Best regards
Peter






From Prometheus001 at gmx.net  Tue Dec 16 00:06:16 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 16 Dec 2008 09:06:16 +0100
Subject: [Freeswitch-users] running custom script with bind_meta_app
In-Reply-To: <11990ade0812150925i33e9cf6ex6a1fe53c423fae9b@mail.gmail.com>
References: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>	<910AFB7A-9CDE-49AE-A72D-7942F8F79DA3@freeswitch.org>
	<11990ade0812150925i33e9cf6ex6a1fe53c423fae9b@mail.gmail.com>
Message-ID: <49476178.7080603@gmx.net>

I use Telegraph with Ruby on Rails to listen on the event socket
interface. With Telegraph you can register on any FS event and get all
channel variables in a hash for further processing. Interactions then
can be done via event_socket intreface.
Telegraph is not finished yet, but for me it was a good and easy point
to start.

Best regards
Peter

Stephen Crosby schrieb:
> I just want to listen for some DTMF sequence while in a conference.
> The conference host should be able to enter the sequence at any time
> (and any number of times) to run a custom script. I've already written
> one in javascript, but I can rewrite it in another language if it's
> easier.
>
> On Mon, Dec 15, 2008 at 7:09 AM, Brian West <brian at freeswitch.org> wrote:
>   
>> What are you wanting to accomplish first?
>> /b
>> On Dec 15, 2008, at 1:32 AM, Stephen Crosby wrote:
>>
>> I'm Stephen Crosby, and I've just started working with freeswitch.
>> It's been great so far.
>>
>> I want to run a custom script inside a conference when a DTMF sequence
>> is entered. I found bind_meta_app and thought it would be perfect, but
>> I can't seem to get it to work. When I dial-in and press *8, I get no
>> debugging output at all. When I press another sequence like *9 for
>> instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf()
>> sofia/external/5593495805 at sip.gafachi.com Ignoring meta digit '9' not
>> mapped. The script I wrote has been tested with "jsrun script.js" from
>> the command line and it does work. I've got the debugging level all
>> the way up and there's just not much for me to go on. Any help would
>> be greatly appreciated.
>>
>> <extension name="gafachi-DID1">
>>  <condition field="destination_number" expression="^xxxxxxxxxxx$">
>>    <action application="answer"/>
>>    <action application="bind_meta_app" data="8 ab s
>> javascript::$${base_dir}/scripts/script.js"/>
>>    <action application="conference" data="3000 at default"/>
>>  </condition>
>> </extension>
>>
>> --Stephen
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From jonas.gauffin at gmail.com  Tue Dec 16 00:07:30 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 16 Dec 2008 09:07:30 +0100
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <49468B9D.5010903@freeswitch.org>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
	<20081215110346.GA12681@jdc.jasonjgw.net>
	<432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>
	<d021d3be0812150801g3d904fb5xe98b281f142a2d95@mail.gmail.com>
	<49468B9D.5010903@freeswitch.org>
Message-ID: <d021d3be0812160007s42a6979bn22f1601c59d862fa@mail.gmail.com>

        <?xml version="1.0" encoding="UTF-8" standalone="no"?>
                <document type="freeswitch/xml">
                  <section name="dialplan" description="ModCurlDialplan">
                    <context name="default">
                      <extension name=" 0236661201 ">
                        <condition field="destination_number"
expression="^0236661201$">
                          <action application="set" data="gate_site_id=1"/>
                          <action application="set"
data="effective_caller_id_number=0812024481"/>
                          <action application="set"
data="gate_external_destination=+46236661201"/>
                          <action application="bridge" data="sofia/default/
0236661201 at sip-corporate2.tele2.se" />
                          <action application="hangup"
data="NORMAL_CLEARING"/>
                        </condition>
                      </extension>
                    </context>
                  </section>
                </document>


That's it. the sofia configs are pretty much default.
On Mon, Dec 15, 2008 at 5:53 PM, Raymond Chandler <
intralanman at freeswitch.org> wrote:

>  posting relevant pieces of your dialplan and sofia configs would probably
> help a bit.
>
> -Ray
>
>
> Jonas Gauffin wrote:
>
> Yeah I know. That's what I'm trying to do, but with the domain name (
> sip-corporate2.tele2.se) instead of the  ip.
>  I'm not sure that the gateway works without authentication and I'm
> wondering if MANDATORY_IE_MISSING means that the gateway wants
> authentication or if it means something else.
> It's the "No Matching gateway found" message that is confusing, since I'm
> not trying to use a registered gateway?
>
> On Mon, Dec 15, 2008 at 4:05 PM, Brian West <brian at freeswitch.org> wrote:
>
>> If you don't need auth you don't need a gateway.  sofia/profile/
>> number at remoteip is all you should need.
>>
>> /b
>>
>> On Dec 15, 2008, at 5:03 AM, Jason White wrote:
>>
>> > On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
>> >> I'm trying to bridge using a non-registered gateway. And I
>> >> get MANDATORY_IE_MISSING back. Why is that?
>> >
>> > Does the gateway allow unauthenticated clients to make calls? If you
>> > obtain a
>> > SIP trace, you'll be able to see whether it's an authentication issue.
>> >
>> > As an aside, it would be an improvement to FreeSWITCH if Sofia
>> > debugging could
>> > be turned on and off within a running FreeSWITCH instance, including
>> > SIP
>> > traces, instead of the administrator's having to restart FreeSWITCH
>> > with
>> > environment variables exported, as is presently required according
>> > to the
>> > wiki.
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fidibus83 at aol.com  Tue Dec 16 01:54:47 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 16 Dec 2008 10:54:47 +0100
Subject: [Freeswitch-users] Zaptel Error!!!
Message-ID: <002101c95f64$56958e60$6445310a@Franzi>

Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

*

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg ?vv:

 

31 channels configured.

 

ioctl(ZT_LOADZONE) failed: Invalid argument

Notice: Configuration file is /etc/zaptel.conf

line 288: Unable to register tone zone 'de'

 

 

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

What do I wrong? Please help me!
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From mszlazak at aol.com  Tue Dec 16 02:02:25 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Tue, 16 Dec 2008 05:02:25 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
Message-ID: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>

I'm making a call internally from a soft phone to an extension that is suppose to bridge the call internally to another application on the same computer. The applications logs indicate that a connection was made but sound is not being passed back from the application through freeswitch to the softphone. There maybe an issue with rtp timing and associated ports but I'm very new at diagnosing this and fixing the problem.

I've attached both a copy of the FS log and an associated pcap file.

It's all on Windows XP. 

Could someone please take a look.

Thanks.
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From jonas.gauffin at gmail.com  Tue Dec 16 02:54:40 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 16 Dec 2008 11:54:40 +0100
Subject: [Freeswitch-users] Bind error
Message-ID: <d021d3be0812160254y1e0ef82cmc83ae914a0b5324a@mail.gmail.com>

I get a bind error for the RTP, can someone be kind and explain why?

2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2388 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000]/[PCMA:8:8000]
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1596 sofia_glue_tech_set_codec()
Set Codec sofia/internal/0236661201 at sip-corporate2.tele2.se PCMA/8000 20 ms
160 samples
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2352 sofia_glue_negotiate_sdp() Set
2833 dtmf payload to 101
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1814 sofia_glue_activate_rtp()
AUDIO RTP [sofia/internal/0236661201 at sip-corporate2.tele2.se] 192.168.1.112
port 17102 -> 130.244.5X.XX port 27354 codec: 8 ms: 20
2008-12-16 10:47:07 [DEBUG] switch_rtp.c:858 switch_rtp_create() Starting
timer [soft] 160 bytes per 20000ms
2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Ring-Ready sofia/internal/0236661201 at sip-corporate2.tele2.se!
2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Pre-Answer sofia/internal/0236661201 at sip-corporate2.tele2.se!
2008-12-16 10:47:07 [DEBUG] switch_channel.c:1615
switch_channel_perform_pre_answer()
sofia/external/0707992871 at 212.151.14X.XXX receive message
[SWITCH_MESSAGE_INDICATE_PROGRESS]
2008-12-16 10:47:07 [INFO] mod_sofia.c:1253 sofia_receive_message() Asked to
send early media by sofia/external/0707992871 at 212.151.14X.XXX
2008-12-16 10:47:07 [DEBUG] switch_channel.c:1585
switch_channel_perform_mark_pre_answered() Send signal
sofia/external/0707992871 at 212.151.14X.XXX [BREAK]
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup()
STUN Success [85.89.7X.XXX]:[33594]
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1814 sofia_glue_activate_rtp()
AUDIO RTP [sofia/external/0707992871 at 212.151.14X.XXX] 192.168.1.112 port
17108 -> 130.244.1X.XXX port 17148 codec: 8 ms: 20
2008-12-16 10:47:07 [DEBUG] switch_rtp.c:858 switch_rtp_create() Starting
timer [soft] 160 bytes per 20000ms

2008-12-16 10:47:07 [ERR] sofia_glue.c:2045 sofia_glue_activate_rtp() AUDIO
RTP REPORTS ERROR: [Bind Error!]

2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2046 sofia_glue_activate_rtp()
Hangup sofia/external/0707992871 at 212.151.14X.XXX [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2008-12-16 10:47:07 [DEBUG] switch_channel.c:1478
switch_channel_perform_hangup() Send signal
sofia/external/0707992871 at 212.151.14X.XXX [KILL]
2008-12-16 10:47:07 [DEBUG] switch_core_session.c:806
switch_core_session_signal_state_change() Send signal
sofia/external/0707992871 at 212.151.14X.XXX [BREAK]
2008-12-16 10:47:07 [INFO] mod_sofia.c:1294 sofia_receive_message() Ring
SDP:
v=0
o=FreeSWITCH 1229387233 1229387234 IN IP4 85.89.7X.XXX
s=FreeSWITCH
c=IN IP4 85.89.7X.XXX
t=0 0
m=audio 33594 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2008-12-16 10:47:07 [NOTICE] mod_sofia.c:1297 sofia_receive_message()
Ring-Ready sofia/external/0707992871 at 212.151.14X.XXX!
2008-12-16 10:47:07 [NOTICE] mod_sofia.c:1297 sofia_receive_message()
Pre-Answer sofia/external/0707992871 at 212.151.14X.XXX!
2008-12-16 10:47:07 [DEBUG] switch_core_session.c:510
switch_core_session_perform_receive_message() Send signal
sofia/external/0707992871 at 212.151.14X.XXX [BREAK]
2008-12-16 10:47:07 [DEBUG] switch_ivr_originate.c:1625
switch_ivr_originate() Originate Resulted in Error Cause: 487
[ORIGINATOR_CANCEL]
2008-12-16 10:47:07 [NOTICE] switch_ivr_originate.c:1666
switch_ivr_originate() Hangup sofia/internal/
0236661201 at sip-corporate2.tele2.se [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2008-12-16 10:47:07 [DEBUG] switch_channel.c:1478
switch_channel_perform_hangup() Send signal sofia/internal/
0236661201 at sip-corporate2.tele2.se [KILL]
2008-12-16 10:47:07 [DEBUG] switch_core_session.c:806
switch_core_session_signal_state_change() Send signal sofia/internal/
0236661201 at sip-corporate2.tele2.se [BREAK]
2008-12-16 10:47:07 [INFO] mod_dptools.c:1869 audio_bridge_function()
Originate Failed.  Cause: ORIGINATOR_CANCEL
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From scott.ellis at novatex.com.au  Tue Dec 16 03:03:43 2008
From: scott.ellis at novatex.com.au (Scott Ellis)
Date: Tue, 16 Dec 2008 22:03:43 +1100
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
Message-ID: <49478B0F.3000802@novatex.com.au>

I have a standard install, and I am trying to get a Pennytel gateway to 
register.

After looking at Wireshark traces of x-lite registering and FreeSwitch 
registering, FreeSwitch is not sending any authentication information 
with the registration request. I am obviously missing something here!

I understand for incoming calls you don't want authentication, but for 
outgoing it is obviously required.

Is there a flag somewhere that I am supposed to set? The file was taken 
from the wiki page, and looks like it was previously tested when using 
the obsolete outbound directory structure.

The following file is in the conf/sip_profiles/external directory.

<include>
  <gateway name="PennyTel">
    <param name="username" value="888917xxxx"/>
    <param name="password" value="xxxxxxx"/>
    <param name="realm" value="sip.pennytel.com"/>
    <param name="proxy" value="sip.pennytel.com"/>
    <param name="register" value="true"/>
    <param name="expire-seconds" value="60"/>
  </gateway>
</include>

Thanks.

Scott



From fidibus83 at aol.com  Tue Dec 16 03:34:49 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 16 Dec 2008 12:34:49 +0100
Subject: [Freeswitch-users]  Openzap ERROR can't dial
Message-ID: <004a01c95f72$4e59aca0$6445310a@Franzi>

Have nobody an idea?

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From helmut.kuper at ewetel.de  Tue Dec 16 04:32:23 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 13:32:23 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493F7148.40705@ewetel.de>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>
	<493F7148.40705@ewetel.de>
Message-ID: <49479FD7.2070200@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

hm have libtiff4 installed. aptitude says  this:

v   libtiff-dev                                                     -
p   libtiff-opengl                                                  -
TIFF manipulation and conversion tools
p   libtiff-tools                                                   -
TIFF manipulation and conversion tools
i A libtiff4                                                        -
Tag Image File Format (TIFF) library
i A libtiff4-dev                                                    -
Tag Image File Format library (TIFF), development files
i A libtiffxx0c2                                                    -
Tag Image File Format (TIFF) library -- C++ interface

so it is there. but still I get this error during compiling:

Compiling mod_fax.c...
In file included from /usr/include/spandsp.h:97,
                 from mod_fax.c:36:
../../../../libs/spandsp/src/spandsp/t38_gateway.h:94: error: expected
specifier-qualifier-list before ?fax_modems_state_t?
../../../../libs/spandsp/src/spandsp/t38_gateway.h:204: error: expected
specifier-qualifier-list before ?t38_non_ecm_buffer_state_t?
In file included from /usr/include/spandsp.h:99,
                 from mod_fax.c:36:
../../../../libs/spandsp/src/spandsp/t31.h:57: error: expected
specifier-qualifier-list before ?fax_modems_state_t?
In file included from /usr/include/spandsp.h:100,
                 from mod_fax.c:36:
../../../../libs/spandsp/src/spandsp/fax.h:52: error: expected
specifier-qualifier-list before ?fax_modems_state_t?

I still have to do a configure and a make in libs/spandsp befor
compiling FS.


regards
helmut
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From vkobashi at ydeasolutions.com.br  Tue Dec 16 04:59:11 2008
From: vkobashi at ydeasolutions.com.br (vinicius)
Date: Tue, 16 Dec 2008 10:59:11 -0200
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <494198F3.10806@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>
	<494198F3.10806@ydeasolutions.com.br>
Message-ID: <4947A61F.6060806@ydeasolutions.com.br>

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From helmut.kuper at ewetel.de  Tue Dec 16 05:06:59 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 14:06:59 +0100
Subject: [Freeswitch-users] mod_ldap
Message-ID: <4947A7F3.90703@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
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Hello,

I updated to latest FS trunk today and tried to compile mod_ldap and got
this errors:


mkdir .libs
ar cru .libs/libldap.a  bind.o open.o result.o error.o compare.o
search.o controls.o messages.o references.o extended.o cyrus.o modify.o
add.o modrdn.o delete.o abandon.o sasl.o sbind.o kbind.o unbind.o
cancel.o filter.o free.o sort.o passwd.o whoami.o getdn.o getentry.o
getattr.o getvalues.o addentry.o request.o os-ip.o url.o sortctrl.o
vlvctrl.o init.o options.o print.o string.o util-int.o schema.o
charray.o tls.o os-local.o dnssrv.o utf-8.o utf-8-conv.o turn.o
groupings.o txn.o ppolicy.o version.o
ranlib .libs/libldap.a
creating libldap.la
(cd .libs && rm -f libldap.la && ln -s ../libldap.la libldap.la)
cc -g -O2 -o apitest apitest.o  ./.libs/libldap.a
/opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/liblber/.libs/liblber.a
../../libraries/liblber/.libs/liblber.a
../../libraries/liblutil/liblutil.a /usr/lib/libsasl2.a -ldl -lresolv
./.libs/libldap.a(os-ip.o): In function `ldap_pvt_is_socket_ready':
/opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194:
warning: `sys_errlist' is deprecated; use `strerror' or `strerror_r' instead
/opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194:
warning: `sys_nerr' is deprecated; use `strerror' or `strerror_r' instead
/usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_close':
(.text+0x348): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
(.text+0x3d0): undefined reference to `db_create'
/usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
(.text+0x453): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_getdata':
(.text+0x7e8): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
(.text+0xa4e): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
(.text+0xafa): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
(.text+0x6c8): undefined reference to `DES_key_sched'
/usr/lib/libsasl2.a(digestmd5.o): In function `init_des':


When I change src/mod/directories/mod_ldap/Makefile line 3 to
"LDAP=openldap-2.4.11" and putting openldap-2.4.11.tgz in libs/ manually
it compiles without error, but with some warnings.


Maybe the openldap-2.3.19 delivered by FS is too old?

regards
helmut
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From carole.olivier at enst.fr  Tue Dec 16 05:07:47 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Tue, 16 Dec 2008 05:07:47 -0800 (PST)
Subject: [Freeswitch-users]  general question about API command
Message-ID: <21032754.post@talk.nabble.com>


Hello,

I have a general question about the API commands. Some of them are not
available in the dialplan like uuid_transfer. I was wondering how to call an
API command without using the CLI. Especially I would be interested in
knowing if there is any way to call them from a phone, I mean bind a key to
an API command.

For instance I would like to transfer both members of a simple call into a
conference by dialing *1. 
I have seen the transfer is possible from the CLI by doing:
api uuid_transfer <uuid> -both 3001
but I do not know how to do it else.

If somebody could give me an insight about the topic it would be great.
Thanks,
Carole








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From mike at jerris.com  Tue Dec 16 05:19:51 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:19:51 -0500
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <002101c95f64$56958e60$6445310a@Franzi>
References: <002101c95f64$56958e60$6445310a@Franzi>
Message-ID: <7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com>

It sounds like there is no de tonezone in the zaptel drivers, but I  
can't imagine thats true.  What version of the drivers do you have  
installed?


On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:

> Hello, I?m a newbie in FS and my English isn?t very good but I try  
> to explain my problem. Hopefully you can understand me. J
>
> I have a Linux-Server with a Digium Wildcard TE110P. I install and  
> configure openzap (PRI/ISDN) and zaptel. But I have an error when I  
> execute ztcfg ?vv:
>
> 31 channels configured.
>
> ioctl(ZT_LOADZONE) failed: Invalid argument
> Notice: Configuration file is /etc/zaptel.conf
> line 288: Unable to register tone zone 'de'
>
>
>
> ZAPTEL.CONF
>
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
> loadzone = de
> defaultzone=de
>
> What do I wrong? Please help me!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Dec 16 05:24:16 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:24:16 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
In-Reply-To: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>
References: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>
Message-ID: <957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com>

If its all local you can also just use:

http://wiki.freeswitch.org/wiki/Bypass_Media

If your still trying to figure it out it could be any number of  
things, but most relating to misconfigured endpoints or freeswitch,  
take a look at the sip trace and make sure everything is using the  
right ip addresses instead of using internal when they should be  
external or the other way around.

Mike

On Dec 16, 2008, at 5:02 AM, mszlazak at aol.com wrote:

> I'm making a call internally from a soft phone to an extension that  
> is suppose to bridge the call internally to another application on  
> the same computer. The applications logs indicate that a connection  
> was made but sound is not being passed back from the application  
> through freeswitch to the softphone. There maybe an issue with rtp  
> timing and associated ports but I'm very new at diagnosing this and  
> fixing the problem.
>
> I've attached both a copy of the FS log and an associated pcap file.
>
> It's all on Windows XP.
>
> Could someone please take a look.
>
> Thanks.
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
> <free.zip>_______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Dec 16 05:25:12 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:25:12 -0500
Subject: [Freeswitch-users] Bind error
In-Reply-To: <d021d3be0812160254y1e0ef82cmc83ae914a0b5324a@mail.gmail.com>
References: <d021d3be0812160254y1e0ef82cmc83ae914a0b5324a@mail.gmail.com>
Message-ID: <85B9BF33-5F2D-4674-9840-069D7D1BB99C@jerris.com>


On Dec 16, 2008, at 5:54 AM, Jonas Gauffin wrote:

> I get a bind error for the RTP, can someone be kind and explain why?
>

Some other program is using the ports already?


From mike at jerris.com  Tue Dec 16 05:26:30 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:26:30 -0500
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
In-Reply-To: <49478B0F.3000802@novatex.com.au>
References: <49478B0F.3000802@novatex.com.au>
Message-ID: <153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>

We send authentication after we get a challenge because on startup we  
need the nonce from them to build the hash in the Auth header properly.

Mike

On Dec 16, 2008, at 6:03 AM, Scott Ellis wrote:

> I have a standard install, and I am trying to get a Pennytel gateway  
> to
> register.
>
> After looking at Wireshark traces of x-lite registering and FreeSwitch
> registering, FreeSwitch is not sending any authentication information
> with the registration request. I am obviously missing something here!
>
> I understand for incoming calls you don't want authentication, but for
> outgoing it is obviously required.
>
> Is there a flag somewhere that I am supposed to set? The file was  
> taken
> from the wiki page, and looks like it was previously tested when using
> the obsolete outbound directory structure.
>
> The following file is in the conf/sip_profiles/external directory.
>
> <include>
>  <gateway name="PennyTel">
>    <param name="username" value="888917xxxx"/>
>    <param name="password" value="xxxxxxx"/>
>    <param name="realm" value="sip.pennytel.com"/>
>    <param name="proxy" value="sip.pennytel.com"/>
>    <param name="register" value="true"/>
>    <param name="expire-seconds" value="60"/>
>  </gateway>
> </include>
>
> Thanks.
>
> Scott
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Tue Dec 16 05:27:10 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:27:10 -0500
Subject: [Freeswitch-users] Openzap ERROR can't dial
In-Reply-To: <004a01c95f72$4e59aca0$6445310a@Franzi>
References: <004a01c95f72$4e59aca0$6445310a@Franzi>
Message-ID: <488DE1CC-8F76-49AE-90EF-ED27263AB134@jerris.com>


On Dec 16, 2008, at 6:34 AM, fidibus83 wrote:

> Have nobody an idea?
>

Idea about what?
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From mike at jerris.com  Tue Dec 16 05:29:19 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:29:19 -0500
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <49479FD7.2070200@ewetel.de>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>
	<493F7148.40705@ewetel.de> <49479FD7.2070200@ewetel.de>
Message-ID: <0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>

The only reason that I can come up with would be if configure is not  
detecting it for whatever reason.  Check the config.log in the main fs  
src dir to see if it mentions the check at all.

MIke

On Dec 16, 2008, at 7:32 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> hm have libtiff4 installed. aptitude says  this:
>
> v   libtiff-dev                                                     -
> p   libtiff-opengl                                                  -
> TIFF manipulation and conversion tools
> p   libtiff-tools                                                   -
> TIFF manipulation and conversion tools
> i A libtiff4                                                        -
> Tag Image File Format (TIFF) library
> i A libtiff4-dev                                                    -
> Tag Image File Format library (TIFF), development files
> i A libtiffxx0c2                                                    -
> Tag Image File Format (TIFF) library -- C++ interface
>
> so it is there. but still I get this error during compiling:
>
> Compiling mod_fax.c...
> In file included from /usr/include/spandsp.h:97,
>                 from mod_fax.c:36:
> ../../../../libs/spandsp/src/spandsp/t38_gateway.h:94: error: expected
> specifier-qualifier-list before ?fax_modems_state_t?
> ../../../../libs/spandsp/src/spandsp/t38_gateway.h:204: error:  
> expected
> specifier-qualifier-list before ?t38_non_ecm_buffer_state_t?
> In file included from /usr/include/spandsp.h:99,
>                 from mod_fax.c:36:
> ../../../../libs/spandsp/src/spandsp/t31.h:57: error: expected
> specifier-qualifier-list before ?fax_modems_state_t?
> In file included from /usr/include/spandsp.h:100,
>                 from mod_fax.c:36:
> ../../../../libs/spandsp/src/spandsp/fax.h:52: error: expected
> specifier-qualifier-list before ?fax_modems_state_t?
>
> I still have to do a configure and a make in libs/spandsp befor
> compiling FS.



From mike at jerris.com  Tue Dec 16 05:32:02 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:32:02 -0500
Subject: [Freeswitch-users] mod_ldap
In-Reply-To: <4947A7F3.90703@ewetel.de>
References: <4947A7F3.90703@ewetel.de>
Message-ID: <818373FA-A5DE-4534-B807-CDE119E9EE42@jerris.com>

Can you please file a bug in jira on this issue, we can get this  
corrected today.
On Dec 16, 2008, at 8:06 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I updated to latest FS trunk today and tried to compile mod_ldap and  
> got
> this errors:
>
>
> mkdir .libs
> ar cru .libs/libldap.a  bind.o open.o result.o error.o compare.o
> search.o controls.o messages.o references.o extended.o cyrus.o  
> modify.o
> add.o modrdn.o delete.o abandon.o sasl.o sbind.o kbind.o unbind.o
> cancel.o filter.o free.o sort.o passwd.o whoami.o getdn.o getentry.o
> getattr.o getvalues.o addentry.o request.o os-ip.o url.o sortctrl.o
> vlvctrl.o init.o options.o print.o string.o util-int.o schema.o
> charray.o tls.o os-local.o dnssrv.o utf-8.o utf-8-conv.o turn.o
> groupings.o txn.o ppolicy.o version.o
> ranlib .libs/libldap.a
> creating libldap.la
> (cd .libs && rm -f libldap.la && ln -s ../libldap.la libldap.la)
> cc -g -O2 -o apitest apitest.o  ./.libs/libldap.a
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/ 
> liblber/.libs/liblber.a
> ../../libraries/liblber/.libs/liblber.a
> ../../libraries/liblutil/liblutil.a /usr/lib/libsasl2.a -ldl -lresolv
> ./.libs/libldap.a(os-ip.o): In function `ldap_pvt_is_socket_ready':
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/ 
> libldap/os-ip.c:194:
> warning: `sys_errlist' is deprecated; use `strerror' or `strerror_r'  
> instead
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/ 
> libldap/os-ip.c:194:
> warning: `sys_nerr' is deprecated; use `strerror' or `strerror_r'  
> instead
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_close':
> (.text+0x348): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
> (.text+0x3d0): undefined reference to `db_create'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
> (.text+0x453): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_getdata':
> (.text+0x7e8): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
> (.text+0xa4e): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
> (.text+0xafa): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
> (.text+0x6c8): undefined reference to `DES_key_sched'
> /usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
>
>
> When I change src/mod/directories/mod_ldap/Makefile line 3 to
> "LDAP=openldap-2.4.11" and putting openldap-2.4.11.tgz in libs/  
> manually
> it compiles without error, but with some warnings.
>
>
> Maybe the openldap-2.3.19 delivered by FS is too old?
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklHp/MACgkQ4tZeNddg3dzjOgCfSExGU7PSWSdAxI4VCZIosrLJ
> 25kAoK0igc53iV1PhWkj5faAVOL78E8B
> =ny1E
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Tue Dec 16 05:48:55 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Dec 2008 07:48:55 -0600
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <21032754.post@talk.nabble.com>
References: <21032754.post@talk.nabble.com>
Message-ID: <191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>

There is a transfer dial plan application also.

There is not really any benefit in blocking the api commands from the
dialplan
apart from the potential for a blocking api call to delay the audio stream
which
you can do at your own risk and use the sleep application with 0ms to flush
the read buffers.

So I guess i can lift that limitation in tree.

see r10790


On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> Hello,
>
> I have a general question about the API commands. Some of them are not
> available in the dialplan like uuid_transfer. I was wondering how to call
> an
> API command without using the CLI. Especially I would be interested in
> knowing if there is any way to call them from a phone, I mean bind a key to
> an API command.
>
> For instance I would like to transfer both members of a simple call into a
> conference by dialing *1.
> I have seen the transfer is possible from the CLI by doing:
> api uuid_transfer <uuid> -both 3001
> but I do not know how to do it else.
>
> If somebody could give me an insight about the topic it would be great.
> Thanks,
> Carole
>
>
>
>
>
>
>
>
> --
> View this message in context:
> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From fidibus83 at aol.com  Tue Dec 16 05:54:01 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 16 Dec 2008 14:54:01 +0100
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com>
References: <002101c95f64$56958e60$6445310a@Franzi>
	<7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com>
Message-ID: <007601c95f85$c1462aa0$6445310a@Franzi>

I have installed zaptel-1.4.11 

I have looked in zonedata.c and there is configured de-tonezone 

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael Jerris
Gesendet: Dienstag, 16. Dezember 2008 14:20
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

It sounds like there is no de tonezone in the zaptel drivers, but I can't
imagine thats true.  What version of the drivers do you have installed?  

 

 

On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:





Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

 

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg ?vv:

 

31 channels configured.

 

ioctl(ZT_LOADZONE) failed: Invalid argument

Notice: Configuration file is /etc/zaptel.conf

line 288: Unable to register tone zone 'de'

 

 

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

What do I wrong? Please help me!

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

=
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From helmut.kuper at ewetel.de  Tue Dec 16 06:35:08 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 15:35:08 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>
	<49479FD7.2070200@ewetel.de>
	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>
Message-ID: <4947BC9C.8040302@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

hm no, there is nothing like libtiff, spandsp or even fax to find in
config.log.

regards
helmut
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1WgAn1dOjzoBgU8Ln6Wri/a53O8rRZOO
=fHh+
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From msc at freeswitch.org  Tue Dec 16 06:37:26 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Dec 2008 06:37:26 -0800
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <007601c95f85$c1462aa0$6445310a@Franzi>
References: <002101c95f64$56958e60$6445310a@Franzi>
	<7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com>
	<007601c95f85$c1462aa0$6445310a@Franzi>
Message-ID: <9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org>

Just a hunch but try removing the spaces in this line:

loadzone=de

Zaptel can be quirky.
-MC

Sent from my iPhone

On Dec 16, 2008, at 5:54 AM, "fidibus83" <fidibus83 at aol.com> wrote:

> I have installed zaptel-1.4.11
>
> I have looked in zonedata.c and there is configured de-tonezone
>
>
>
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] Im Auftrag von Michael Jerris
> Gesendet: Dienstag, 16. Dezember 2008 14:20
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Zaptel Error!!!
>
>
>
> It sounds like there is no de tonezone in the zaptel drivers, but I  
> can't imagine thats true.  What version of the drivers do you have  
> installed?
>
>
>
>
>
> On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:
>
>
>
>
> Hello, I?m a newbie in FS and my English isn?t very good but I  
> try to explain my problem. Hopefully you can understand me. J
>
>
>
> I have a Linux-Server with a Digium Wildcard TE110P. I install and  
> configure openzap (PRI/ISDN) and zaptel. But I have an error when I  
> execute ztcfg ?vv:
>
>
>
> 31 channels configured.
>
>
>
> ioctl(ZT_LOADZONE) failed: Invalid argument
>
> Notice: Configuration file is /etc/zaptel.conf
>
> line 288: Unable to register tone zone 'de'
>
>
>
>
>
>
>
> ZAPTEL.CONF
>
>
>
> span=1,1,0,ccs,hdb3,crc4
>
> bchan=1-15,17-31
>
> dchan=16
>
> loadzone = de
>
> defaultzone=de
>
>
>
> What do I wrong? Please help me!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> =
> _______________________________________________
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> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From carole.olivier at enst.fr  Tue Dec 16 06:43:32 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Tue, 16 Dec 2008 06:43:32 -0800 (PST)
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
Message-ID: <21033993.post@talk.nabble.com>


Thanks for this answer. 
Just a question so that everything is clear: there is no command to bridge
directly a call and both legs into a conference? We have to transfer the
call to another extension and from there create the conference isn't?

I used the command transfer but I still have a problem. I do the following
and it did not work: in my dialplan I write:
<action application="bind_meta_app" data="1 ab s transfer::3333"/>

The extension 3333 works fine.

However, nothing happens when I press *1.
Do you have an idea where am I wrong?

(A subsidiary question: will both legs be transferred to the extension 3333
or just the one which press *1? is there a way to transfer both together?)


Thanks a lot,
Carole


Anthony Minessale-2 wrote:
> 
> There is a transfer dial plan application also.
> 
> There is not really any benefit in blocking the api commands from the
> dialplan
> apart from the potential for a blocking api call to delay the audio stream
> which
> you can do at your own risk and use the sleep application with 0ms to
> flush
> the read buffers.
> 
> So I guess i can lift that limitation in tree.
> 
> see r10790
> 
> 
> On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr> wrote:
> 
>>
>> Hello,
>>
>> I have a general question about the API commands. Some of them are not
>> available in the dialplan like uuid_transfer. I was wondering how to call
>> an
>> API command without using the CLI. Especially I would be interested in
>> knowing if there is any way to call them from a phone, I mean bind a key
>> to
>> an API command.
>>
>> For instance I would like to transfer both members of a simple call into
>> a
>> conference by dialing *1.
>> I have seen the transfer is possible from the CLI by doing:
>> api uuid_transfer <uuid> -both 3001
>> but I do not know how to do it else.
>>
>> If somebody could give me an insight about the topic it would be great.
>> Thanks,
>> Carole
>>
>>
>>
>>
>>
>>
>>
>>
>> --
>> View this message in context:
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From helmut.kuper at ewetel.de  Tue Dec 16 06:45:34 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 15:45:34 +0100
Subject: [Freeswitch-users] OpenZAP:  "Received unhandled message 125"
In-Reply-To: <EBD35DE7-A6C9-4072-AC8E-F2D49E7098C8@freeswitch.org>
References: <492ED886.9020805@gmx.net>
	<EBD35DE7-A6C9-4072-AC8E-F2D49E7098C8@freeswitch.org>
Message-ID: <4947BF0E.50702@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I get that "CRIT" error on my q931 pri as well. I found that the message
is a Q931 status message from remote end and I guess it only will tell
you the call state of your outgoing call at the remote end to make sure
you are in same state. So it is a nice to have to get this mesage
decoded for debugging reasons.

regards
helmut



Am 27.11.2008 19:02, schrieb Michael S Collins:
> You can ignore this one for now. Eventually this will be handled but  
> it shouldn't affect your calls. I've been ignoring it for six months. :)
> 
> -MC
> 
> Sent from my iPhone
> 
> On Nov 27, 2008, at 9:27 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> 
>> I have installed OpenZAP with a TE220 card and EuroISDN.
>>
>> When I bridge an outgoing call I get a "Received unhandled message  
>> 125"
>> Any idea what that means? As far as I know there is no <external>  
>> result
>> code 125 defined in the ISDN protocol.
>>
>> Best regards
>> Peter
>>
>> See the following logs.
>>
>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14
>> --- 
>> --- 
>> --- 
>> --- 
>> --------------------------------------------------------------------
>> [08 02 80 03 7d 08 04 82 e3 98 28 14 01 01]
>>
>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Received message from Q.921
>> (ind 4, tei 0, size 18)
>> MesType: 125, CRVFlag 1 (Terminator), CRV 3 (Dialect: 2)
>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Sending message to Layer4
>> (size: 114)
>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I  
>> got
>> an event! Type:[7d] Size:[114] CRV: 3 (0x3, CTX: Terminator)
>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan
>> 70d230 (1:1) source isdn_data->channels_local_crv[0x3]
>> 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received
>> unhandled message 125 (0x7d)
>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.921() Sending frame
>>
>>
>>> oz list
>> API CALL [oz(list)] output:
>> +OK
>> span: 1 (span1)
>> type: isdn
>> chan_count: 31
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>> +OK
>> span: 2 (span2)
>> type: isdn
>> chan_count: 31
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
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WKkAnjkDOXKrgFKQNHyeSyvw6rI2JW0U
=Ax8Y
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From fidibus83 at aol.com  Tue Dec 16 06:49:17 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 16 Dec 2008 15:49:17 +0100
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org>
References: <002101c95f64$56958e60$6445310a@Franzi><7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com><007601c95f85$c1462aa0$6445310a@Franzi>
	<9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org>
Message-ID: <009f01c95f8d$7918ddb0$6445310a@Franzi>

It?s already the same error.

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael S Collins
Gesendet: Dienstag, 16. Dezember 2008 15:37
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

Just a hunch but try removing the spaces in this line:

 

loadzone=de

Zaptel can be quirky. 

-MC


Sent from my iPhone


On Dec 16, 2008, at 5:54 AM, "fidibus83" <fidibus83 at aol.com> wrote:

I have installed zaptel-1.4.11 

I have looked in zonedata.c and there is configured de-tonezone 

 


  _____  


Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael Jerris
Gesendet: Dienstag, 16. Dezember 2008 14:20
An:  <mailto:freeswitch-users at lists.freeswitch.org>
freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

It sounds like there is no de tonezone in the zaptel drivers, but I can't
imagine thats true.  What version of the drivers do you have installed?  

 

 

On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:








Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

 

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg ?vv:

 

31 channels configured.

 

ioctl(ZT_LOADZONE) failed: Invalid argument

Notice: Configuration file is /etc/zaptel.conf

line 288: Unable to register tone zone 'de'

 

 

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

What do I wrong? Please help me!

_______________________________________________
Freeswitch-users mailing list
 <mailto:Freeswitch-users at lists.freeswitch.org>
Freeswitch-users at lists.freeswitch.org
 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>
http://lists.freeswitch.org/mailman/options/freeswitch-users
 <http://www.freeswitch.org> http://www.freeswitch.org

 

= 

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>
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=
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From helmut.kuper at ewetel.de  Tue Dec 16 06:58:21 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 15:58:21 +0100
Subject: [Freeswitch-users] mod_ldap
In-Reply-To: <818373FA-A5DE-4534-B807-CDE119E9EE42@jerris.com>
References: <4947A7F3.90703@ewetel.de>
	<818373FA-A5DE-4534-B807-CDE119E9EE42@jerris.com>
Message-ID: <4947C20D.4010704@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

k did it: http://jira.freeswitch.org/browse/MDXMLINT-44

Didn't know where to put it, so I put it to "XML interfaces"

regards
helmut


Am 16.12.2008 14:32, schrieb Michael Jerris:
> Can you please file a bug in jira on this issue, we can get this  
> corrected today.
> On Dec 16, 2008, at 8:06 AM, Helmut Kuper wrote:
>
-----BEGIN PGP SIGNATURE-----
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iEYEARECAAYFAklHwgwACgkQ4tZeNddg3dyDfQCgq/VLYtvat1P8r0dCMgAVlPUS
zdcAmgOEylXNMMfbUStlj6w0CqMbHPPP
=5dcw
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From mike at jerris.com  Tue Dec 16 07:00:31 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 10:00:31 -0500
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <4947BC9C.8040302@ewetel.de>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>
	<49479FD7.2070200@ewetel.de>
	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>
	<4947BC9C.8040302@ewetel.de>
Message-ID: <E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>

possibly you have older code that was not bootstrapped again when we  
added spandsp?  Try update, bootstrap, configure again and see if its  
there after.

Mike

On Dec 16, 2008, at 9:35 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> hm no, there is nothing like libtiff, spandsp or even fax to find in
> config.log.
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklHvJsACgkQ4tZeNddg3dwsOwCeI1hVTC3CeW/SLiuHn0g6VTYW
> 1WgAn1dOjzoBgU8Ln6Wri/a53O8rRZOO
> =fHh+
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Tue Dec 16 07:02:10 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 10:02:10 -0500
Subject: [Freeswitch-users] OpenZAP:  "Received unhandled message 125"
In-Reply-To: <4947BF0E.50702@ewetel.de>
References: <492ED886.9020805@gmx.net>
	<EBD35DE7-A6C9-4072-AC8E-F2D49E7098C8@freeswitch.org>
	<4947BF0E.50702@ewetel.de>
Message-ID: <8521EE9F-DBE2-40B5-B188-FE124DA77002@jerris.com>

That is just a message we don't handle yet, it will be added in the  
future.

Mike

On Dec 16, 2008, at 9:45 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I get that "CRIT" error on my q931 pri as well. I found that the  
> message
> is a Q931 status message from remote end and I guess it only will tell
> you the call state of your outgoing call at the remote end to make  
> sure
> you are in same state. So it is a nice to have to get this mesage
> decoded for debugging reasons.
>>> 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34()  
>>> Received
>>> unhandled message 125 (0x7d)
>>>


From vkobashi at ydeasolutions.com.br  Tue Dec 16 07:08:24 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Tue, 16 Dec 2008 13:08:24 -0200
Subject: [Freeswitch-users] mod_ldap
In-Reply-To: <4947A7F3.90703@ewetel.de>
References: <4947A7F3.90703@ewetel.de>
Message-ID: <4947C468.7050100@ydeasolutions.com.br>

yes it is, you need to hack fs to get the latest version of openldap for you

Helmut Kuper escreveu:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I updated to latest FS trunk today and tried to compile mod_ldap and got
> this errors:
>
>
> mkdir .libs
> ar cru .libs/libldap.a  bind.o open.o result.o error.o compare.o
> search.o controls.o messages.o references.o extended.o cyrus.o modify.o
> add.o modrdn.o delete.o abandon.o sasl.o sbind.o kbind.o unbind.o
> cancel.o filter.o free.o sort.o passwd.o whoami.o getdn.o getentry.o
> getattr.o getvalues.o addentry.o request.o os-ip.o url.o sortctrl.o
> vlvctrl.o init.o options.o print.o string.o util-int.o schema.o
> charray.o tls.o os-local.o dnssrv.o utf-8.o utf-8-conv.o turn.o
> groupings.o txn.o ppolicy.o version.o
> ranlib .libs/libldap.a
> creating libldap.la
> (cd .libs && rm -f libldap.la && ln -s ../libldap.la libldap.la)
> cc -g -O2 -o apitest apitest.o  ./.libs/libldap.a
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/liblber/.libs/liblber.a
> ../../libraries/liblber/.libs/liblber.a
> ../../libraries/liblutil/liblutil.a /usr/lib/libsasl2.a -ldl -lresolv
> ./.libs/libldap.a(os-ip.o): In function `ldap_pvt_is_socket_ready':
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194:
> warning: `sys_errlist' is deprecated; use `strerror' or `strerror_r' instead
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194:
> warning: `sys_nerr' is deprecated; use `strerror' or `strerror_r' instead
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_close':
> (.text+0x348): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
> (.text+0x3d0): undefined reference to `db_create'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
> (.text+0x453): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_getdata':
> (.text+0x7e8): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
> (.text+0xa4e): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
> (.text+0xafa): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
> (.text+0x6c8): undefined reference to `DES_key_sched'
> /usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
>
>
> When I change src/mod/directories/mod_ldap/Makefile line 3 to
> "LDAP=openldap-2.4.11" and putting openldap-2.4.11.tgz in libs/ manually
> it compiles without error, but with some warnings.
>
>
> Maybe the openldap-2.3.19 delivered by FS is too old?
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
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> 25kAoK0igc53iV1PhWkj5faAVOL78E8B
> =ny1E
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From anthony.minessale at gmail.com  Tue Dec 16 07:17:29 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Dec 2008 09:17:29 -0600
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <21033993.post@talk.nabble.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
Message-ID: <191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>

it should work with
<action application="bind_meta_app" data="1 ab s transfer::-both 3333"/>

make sure you have debug log on (press f8) to see if there are any issues.


On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> Thanks for this answer.
> Just a question so that everything is clear: there is no command to bridge
> directly a call and both legs into a conference? We have to transfer the
> call to another extension and from there create the conference isn't?
>
> I used the command transfer but I still have a problem. I do the following
> and it did not work: in my dialplan I write:
> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
>
> The extension 3333 works fine.
>
> However, nothing happens when I press *1.
> Do you have an idea where am I wrong?
>
> (A subsidiary question: will both legs be transferred to the extension 3333
> or just the one which press *1? is there a way to transfer both together?)
>
>
> Thanks a lot,
> Carole
>
>
> Anthony Minessale-2 wrote:
> >
> > There is a transfer dial plan application also.
> >
> > There is not really any benefit in blocking the api commands from the
> > dialplan
> > apart from the potential for a blocking api call to delay the audio
> stream
> > which
> > you can do at your own risk and use the sleep application with 0ms to
> > flush
> > the read buffers.
> >
> > So I guess i can lift that limitation in tree.
> >
> > see r10790
> >
> >
> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
> wrote:
> >
> >>
> >> Hello,
> >>
> >> I have a general question about the API commands. Some of them are not
> >> available in the dialplan like uuid_transfer. I was wondering how to
> call
> >> an
> >> API command without using the CLI. Especially I would be interested in
> >> knowing if there is any way to call them from a phone, I mean bind a key
> >> to
> >> an API command.
> >>
> >> For instance I would like to transfer both members of a simple call into
> >> a
> >> conference by dialing *1.
> >> I have seen the transfer is possible from the CLI by doing:
> >> api uuid_transfer <uuid> -both 3001
> >> but I do not know how to do it else.
> >>
> >> If somebody could give me an insight about the topic it would be great.
> >> Thanks,
> >> Carole
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> --
> >> View this message in context:
> >>
> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From helmut.kuper at ewetel.de  Tue Dec 16 07:39:55 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 16:39:55 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
Message-ID: <4947CBCB.8060204@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

H Anthony,

I have same problem on each 7th outgoing call. Then a sort of timeout
occurs. During that time, no incomming call is possible.

regards
helmut


Am 23.09.2008 16:13, schrieb Anthony Minessale:
> I have moved your issue to  http://jira.freeswitch.org/browse/OPENZAP-18
> since it's an openzap issue and not a build system issue.
> 
> Please pay attention to the emails you receive regarding the issue and
> promptly reply to any comments.
> 
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iEYEARECAAYFAklHy8sACgkQ4tZeNddg3dynXgCeLco/4Pmu3hFVVFHP/glucTtS
o4oAoIo1q2bA5ItCwGx4y9kZsy1KPWlp
=Rv7D
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From msc at freeswitch.org  Tue Dec 16 07:49:46 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Dec 2008 07:49:46 -0800
Subject: [Freeswitch-users] OpenZAP:  "Received unhandled message 125"
In-Reply-To: <4947BF0E.50702@ewetel.de>
References: <492ED886.9020805@gmx.net>
	<EBD35DE7-A6C9-4072-AC8E-F2D49E7098C8@freeswitch.org>
	<4947BF0E.50702@ewetel.de>
Message-ID: <CC1175B5-5B44-4C65-A836-D6079BA6F358@freeswitch.org>

It's on the list of things to do. Since there is no bounty for getting  
the pri stack finished it's a little bit lower priority. However, it  
is definitely being worked on.

-MC

Sent from my iPhone

On Dec 16, 2008, at 6:45 AM, Helmut Kuper <helmut.kuper at ewetel.de>  
wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I get that "CRIT" error on my q931 pri as well. I found that the  
> message
> is a Q931 status message from remote end and I guess it only will tell
> you the call state of your outgoing call at the remote end to make  
> sure
> you are in same state. So it is a nice to have to get this mesage
> decoded for debugging reasons.
>
> regards
> helmut
>
>
>
> Am 27.11.2008 19:02, schrieb Michael S Collins:
>> You can ignore this one for now. Eventually this will be handled but
>> it shouldn't affect your calls. I've been ignoring it for six  
>> months. :)
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Nov 27, 2008, at 9:27 AM, Peter P GMX <Prometheus001 at gmx.net>  
>> wrote:
>>
>>> I have installed OpenZAP with a TE220 card and EuroISDN.
>>>
>>> When I bridge an outgoing call I get a "Received unhandled message
>>> 125"
>>> Any idea what that means? As far as I know there is no <external>
>>> result
>>> code 125 defined in the ISDN protocol.
>>>
>>> Best regards
>>> Peter
>>>
>>> See the following logs.
>>>
>>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23()  
>>> READ 14
>>> ---
>>> ---
>>> ---
>>> ---
>>> --------------------------------------------------------------------
>>> [08 02 80 03 7d 08 04 82 e3 98 28 14 01 01]
>>>
>>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Received message from Q. 
>>> 921
>>> (ind 4, tei 0, size 18)
>>> MesType: 125, CRVFlag 1 (Terminator), CRV 3 (Dialect: 2)
>>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Sending message to Layer4
>>> (size: 114)
>>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I
>>> got
>>> an event! Type:[7d] Size:[114] CRV: 3 (0x3, CTX: Terminator)
>>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan
>>> 70d230 (1:1) source isdn_data->channels_local_crv[0x3]
>>> 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34()  
>>> Received
>>> unhandled message 125 (0x7d)
>>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.921() Sending frame
>>>
>>>
>>>> oz list
>>> API CALL [oz(list)] output:
>>> +OK
>>> span: 1 (span1)
>>> type: isdn
>>> chan_count: 31
>>> dialplan: XML
>>> context: default
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options none
>>> +OK
>>> span: 2 (span2)
>>> type: isdn
>>> chan_count: 31
>>> dialplan: XML
>>> context: default
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options none
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklHvw0ACgkQ4tZeNddg3dyAnACfaXoHklB3xfaaQXTB+raTyASV
> WKkAnjkDOXKrgFKQNHyeSyvw6rI2JW0U
> =Ax8Y
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Tue Dec 16 07:52:41 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Dec 2008 07:52:41 -0800
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <009f01c95f8d$7918ddb0$6445310a@Franzi>
References: <002101c95f64$56958e60$6445310a@Franzi><7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com><007601c95f85$c1462aa0$6445310a@Franzi>
	<9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org>
	<009f01c95f8d$7918ddb0$6445310a@Franzi>
Message-ID: <B4F10400-D789-4507-8C06-F2B7F79ECED7@freeswitch.org>

Thanks for trying. You might want to clean out your zaptel install and  
do a complete reinstall.

-MC

Sent from my iPhone

On Dec 16, 2008, at 6:49 AM, "fidibus83" <fidibus83 at aol.com> wrote:

> It?s already the same error.
>
>
>
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] Im Auftrag von Michael S Collins
> Gesendet: Dienstag, 16. Dezember 2008 15:37
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Zaptel Error!!!
>
>
>
> Just a hunch but try removing the spaces in this line:
>
>
>
> loadzone=de
>
> Zaptel can be quirky.
>
> -MC
>
>
> Sent from my iPhone
>
>
> On Dec 16, 2008, at 5:54 AM, "fidibus83" <fidibus83 at aol.com> wrote:
>
>> I have installed zaptel-1.4.11
>>
>> I have looked in zonedata.c and there is configured de-tonezone
>>
>>
>>
>> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
>> ] Im Auftrag von Michael Jerris
>> Gesendet: Dienstag, 16. Dezember 2008 14:20
>> An: freeswitch-users at lists.freeswitch.org
>> Betreff: Re: [Freeswitch-users] Zaptel Error!!!
>>
>>
>>
>> It sounds like there is no de tonezone in the zaptel drivers, but I  
>> can't imagine thats true.  What version of the drivers do you have  
>> installed?
>>
>>
>>
>>
>>
>> On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:
>>
>>
>>
>>
>>
>> Hello, I?m a newbie in FS and my English isn?t very good but I  
>> try to explain my problem. Hopefully you can understand me. J
>>
>>
>>
>> I have a Linux-Server with a Digium Wildcard TE110P. I install and  
>> configure openzap (PRI/ISDN) and zaptel. But I have an error when I  
>> execute ztcfg ?vv:
>>
>>
>>
>> 31 channels configured.
>>
>>
>>
>> ioctl(ZT_LOADZONE) failed: Invalid argument
>>
>> Notice: Configuration file is /etc/zaptel.conf
>>
>> line 288: Unable to register tone zone 'de'
>>
>>
>>
>>
>>
>>
>>
>> ZAPTEL.CONF
>>
>>
>>
>> span=1,1,0,ccs,hdb3,crc4
>>
>> bchan=1-15,17-31
>>
>> dchan=16
>>
>> loadzone = de
>>
>> defaultzone=de
>>
>>
>>
>> What do I wrong? Please help me!
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> =
>>
>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> = h.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> = y> =
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Tue Dec 16 08:04:32 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Dec 2008 08:04:32 -0800
Subject: [Freeswitch-users] Speed Dial Emulation
Message-ID: <686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>

Do you need something just for one extension? Or system wide?

If it's system wide then all you need is an extension that matches a  
condition like this:

<condition field="destination_number" expression="^1$">

Anyone who dials just a single digit 1 will go to this Dialplan entry.  
 From there you can have it bridge to whatever endpoint you like.

-MC

Sent from my iPhone

On Dec 15, 2008, at 7:45 AM, jflowers <jflowers at ezo.net> wrote:

>
> How do I emulate a speed dial setup.  That is, from extension 1003 I  
> dial
> just a 1 ( or 2, or 3 etc.) and nothing else and freeswitch dials a  
> PSTN
> number.  Is there software to do this?
>
>
> -- 
> View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21016167.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Dec 16 08:10:42 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 10:10:42 -0600
Subject: [Freeswitch-users] Speed Dial Emulation
In-Reply-To: <686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>
References: <686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>
Message-ID: <154FFEA4-3E17-4570-B8D6-E54ED2D4A028@freeswitch.org>

Tie that to the db application and you'll have it.

/b

On Dec 16, 2008, at 10:04 AM, Michael S Collins wrote:

> Do you need something just for one extension? Or system wide?
>
> If it's system wide then all you need is an extension that matches a
> condition like this:
>
> <condition field="destination_number" expression="^1$">
>
> Anyone who dials just a single digit 1 will go to this Dialplan entry.
> From there you can have it bridge to whatever endpoint you like.
>
> -MC
>
> Sent from my iPhone



From helmut.kuper at ewetel.de  Tue Dec 16 08:44:17 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 17:44:17 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <4947CBCB.8060204@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de>
Message-ID: <4947DAE1.1050706@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello again,

hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
about  network congestion while mine is about a pri blocking timeout on
every 7th outgoing call.
So, shall I open a seperate bug on jira?

regards
helmut



Am 16.12.2008 16:39, schrieb Helmut Kuper:
> H Anthony,
> 
> I have same problem on each 7th outgoing call. Then a sort of timeout
> occurs. During that time, no incomming call is possible.
> 
> regards
> helmut
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=H2ic
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From msc at freeswitch.org  Tue Dec 16 09:00:11 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Dec 2008 09:00:11 -0800
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <4947DAE1.1050706@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
Message-ID: <87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>

Yes please. Also, if you can attach some debug information, preferably
reproducing the symptoms with the full debug turned on (F8) and attach
it as a file that would assist with the research.

-MC

On Tue, Dec 16, 2008 at 8:44 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello again,
>
> hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
> about  network congestion while mine is about a pri blocking timeout on
> every 7th outgoing call.
> So, shall I open a seperate bug on jira?
>
> regards
> helmut
>
>
>
> Am 16.12.2008 16:39, schrieb Helmut Kuper:
>> H Anthony,
>>
>> I have same problem on each 7th outgoing call. Then a sort of timeout
>> occurs. During that time, no incomming call is possible.
>>
>> regards
>> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
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> 0ugAnj1kX0w1wI45zjeouVz7VMA3d1UN
> =H2ic
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From helmut.kuper at ewetel.de  Tue Dec 16 08:59:50 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 17:59:50 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>	<49479FD7.2070200@ewetel.de>	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>	<4947BC9C.8040302@ewetel.de>
	<E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>
Message-ID: <4947DE86.2010904@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Mike,

very likely. Currently I update an recompile FS serveral times a day :/

I did a "svn update", deleted the libs directory, did a bootstrap and
configure. This time in config.log I have a hint about spandsp :)

Buuuuuut ... I get this now :

making install mod_fax
Compiling mod_fax.c...
Creating mod_fax.so...
quiet_libtool: link: cannot find the library
`../../../../libs/spandsp/src/libspandsp.la' or unhandled argument
`../../../../libs/spandsp/src/libspandsp.la'
make[5]: *** [mod_fax.so] Error 1
make[4]: *** [install] Error 1
make[3]: *** [mod_fax-install] Error 1
make[2]: *** [install-recursive] Error 1
Making install in build

This time I had only to do a "make" in libs/spandsp directory to get FS
compiled.


regards
Helmut

Am 16.12.2008 16:00, schrieb Michael Jerris:
> possibly you have older code that was not bootstrapped again when we  
> added spandsp?  Try update, bootstrap, configure again and see if its  
> there after.
> 
> Mike
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-----END PGP SIGNATURE-----


From brian at freeswitch.org  Tue Dec 16 09:03:50 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 11:03:50 -0600
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <4947DE86.2010904@ewetel.de>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>	<49479FD7.2070200@ewetel.de>	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>	<4947BC9C.8040302@ewetel.de>
	<E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>
	<4947DE86.2010904@ewetel.de>
Message-ID: <D4E0568C-D219-4B3E-B964-F0C641DA805B@freeswitch.org>

You deleted the libs folder?  That might be why.

/b

On Dec 16, 2008, at 10:59 AM, Helmut Kuper wrote:

> I did a "svn update", deleted the libs directory, did a bootstrap and
> configure. This time in config.log I have a hint about spandsp :)



From helmut.kuper at ewetel.de  Tue Dec 16 09:14:46 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 18:14:46 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <D4E0568C-D219-4B3E-B964-F0C641DA805B@freeswitch.org>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>	<49479FD7.2070200@ewetel.de>	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>	<4947BC9C.8040302@ewetel.de>	<E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>	<4947DE86.2010904@ewetel.de>
	<D4E0568C-D219-4B3E-B964-F0C641DA805B@freeswitch.org>
Message-ID: <4947E206.7080504@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

oh sorry m8. first I deleted libs dir, then I did a svn update, so ne
fresh libs dir is installed  ... it's late ...  sorry


helmut


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From anthony.minessale at gmail.com  Tue Dec 16 09:35:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Dec 2008 11:35:51 -0600
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
Message-ID: <191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>

It looks like the other side is not responding at all to the setup message
so the span is locked waiting to see which channel should be assigned.
There is a mutex here stopping the span from using any more channels until
it gets a reply to that setup message.

We could either lower the 60 second timeout to a smaller val, unlock the
mutex before we make the call (possible race condition) or figure out why
you are getting no reply at all from your setup message.


On Tue, Dec 16, 2008 at 11:00 AM, Michael Collins <msc at freeswitch.org>wrote:

> Yes please. Also, if you can attach some debug information, preferably
> reproducing the symptoms with the full debug turned on (F8) and attach
> it as a file that would assist with the research.
>
> -MC
>
> On Tue, Dec 16, 2008 at 8:44 AM, Helmut Kuper <helmut.kuper at ewetel.de>
> wrote:
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: SHA1
> >
> > Hello again,
> >
> > hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
> > about  network congestion while mine is about a pri blocking timeout on
> > every 7th outgoing call.
> > So, shall I open a seperate bug on jira?
> >
> > regards
> > helmut
> >
> >
> >
> > Am 16.12.2008 16:39, schrieb Helmut Kuper:
> >> H Anthony,
> >>
> >> I have same problem on each 7th outgoing call. Then a sort of timeout
> >> occurs. During that time, no incomming call is possible.
> >>
> >> regards
> >> helmut
> > -----BEGIN PGP SIGNATURE-----
> > Version: GnuPG v1.4.9 (MingW32)
> >
> > iEYEARECAAYFAklH2uEACgkQ4tZeNddg3dxZ1ACgs5wZKty2kvzbqa57NONhJ67R
> > 0ugAnj1kX0w1wI45zjeouVz7VMA3d1UN
> > =H2ic
> > -----END PGP SIGNATURE-----
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From helmut.kuper at ewetel.de  Tue Dec 16 09:56:02 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 18:56:02 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>	<4947CBCB.8060204@ewetel.de>
	<4947DAE1.1050706@ewetel.de>	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
Message-ID: <4947EBB2.7030401@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

yes I agree, but hey, the other side is a matured EWSD unlikely that
there is an issue left on protocol level after 30 years. I check that
tomorrow.

regards
helmut

Am 16.12.2008 18:35, schrieb Anthony Minessale:
> It looks like the other side is not responding at all to the setup message
> so the span is locked waiting to see which channel should be assigned.
> There is a mutex here stopping the span from using any more channels until
> it gets a reply to that setup message.
> 
> We could either lower the 60 second timeout to a smaller val, unlock the
> mutex before we make the call (possible race condition) or figure out why
> you are getting no reply at all from your setup message.
> 
> 
> On Tue, Dec 16, 2008 at 11:00 AM, Michael Collins <msc at freeswitch.org>wrote:
> 
>> Yes please. Also, if you can attach some debug information, preferably
>> reproducing the symptoms with the full debug turned on (F8) and attach
>> it as a file that would assist with the research.
>>
>> -MC
>>
>> On Tue, Dec 16, 2008 at 8:44 AM, Helmut Kuper <helmut.kuper at ewetel.de>
>> wrote:
> Hello again,
> 
> hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
> about  network congestion while mine is about a pri blocking timeout on
> every 7th outgoing call.
> So, shall I open a seperate bug on jira?
> 
> regards
> helmut
> 
> 
> 
> Am 16.12.2008 16:39, schrieb Helmut Kuper:
>>>>> H Anthony,
>>>>>
>>>>> I have same problem on each 7th outgoing call. Then a sort of timeout
>>>>> occurs. During that time, no incomming call is possible.
>>>>>
>>>>> regards
>>>>> helmut
>>>
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
>>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>

> ------------------------------------------------------------------------

> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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=Q1vB
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From anthony.minessale at gmail.com  Tue Dec 16 10:06:56 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Dec 2008 12:06:56 -0600
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <4947EBB2.7030401@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
	<4947EBB2.7030401@ewetel.de>
Message-ID: <191c3a030812161006s3ceac738m184304f47ca7eb87@mail.gmail.com>

yes I doubt your EWSD is doing anything wrong, but the communication channel
must be interrupted somehow to never get any replies so yes we can try to
figure out what has gone wrong in the communication layer.



On Tue, Dec 16, 2008 at 11:56 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> yes I agree, but hey, the other side is a matured EWSD unlikely that
> there is an issue left on protocol level after 30 years. I check that
> tomorrow.
>
> regards
> helmut
>
> Am 16.12.2008 18:35, schrieb Anthony Minessale:
> > It looks like the other side is not responding at all to the setup
> message
> > so the span is locked waiting to see which channel should be assigned.
> > There is a mutex here stopping the span from using any more channels
> until
> > it gets a reply to that setup message.
> >
> > We could either lower the 60 second timeout to a smaller val, unlock the
> > mutex before we make the call (possible race condition) or figure out why
> > you are getting no reply at all from your setup message.
> >
> >
> > On Tue, Dec 16, 2008 at 11:00 AM, Michael Collins <msc at freeswitch.org
> >wrote:
> >
> >> Yes please. Also, if you can attach some debug information, preferably
> >> reproducing the symptoms with the full debug turned on (F8) and attach
> >> it as a file that would assist with the research.
> >>
> >> -MC
> >>
> >> On Tue, Dec 16, 2008 at 8:44 AM, Helmut Kuper <helmut.kuper at ewetel.de>
> >> wrote:
> > Hello again,
> >
> > hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
> > about  network congestion while mine is about a pri blocking timeout on
> > every 7th outgoing call.
> > So, shall I open a seperate bug on jira?
> >
> > regards
> > helmut
> >
> >
> >
> > Am 16.12.2008 16:39, schrieb Helmut Kuper:
> >>>>> H Anthony,
> >>>>>
> >>>>> I have same problem on each 7th outgoing call. Then a sort of timeout
> >>>>> occurs. During that time, no incomming call is possible.
> >>>>>
> >>>>> regards
> >>>>> helmut
> >>>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> >>>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
>
> > ------------------------------------------------------------------------
>
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklH67IACgkQ4tZeNddg3dyuyQCeJYjTLdCCWjE3pIrFnwbw/cY2
> ZP4AoIFVFdWEPuIcy0Lbi5yhZ4/4sT/5
> =Q1vB
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Dec 16 10:13:57 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 12:13:57 -0600
Subject: [Freeswitch-users] FreeSWITCH Infrastructure / Happy Holidays!
Message-ID: <80B00891-6226-44DE-9430-6C3CC2A22D3F@freeswitch.org>

FreeSWITCers,
	As the project grows I felt the need for the project to own the  
critical infrastructure hosting things like SVN, Jira, Fisheye and  
various other things we as a project use.  I have personally paid for  
two servers that Bandwidth.com is going to be colocating for the  
project in their data center.  Teliax paid for and shipped me the two  
3ware raid1 cards for the machines which was a great help.  The  
servers are going into production after the first of the year and in  
the process we are rebuilding everything to allow for single sign on  
for every service on the FreeSWITCH project.

The servers are in the hands of bandwidth.com to be racked up so we  
can start deployment, as you can see we have our work cut out for us  
with the integration and rebuild of our entire infrastructure.  ;)

If you wish to pitch in my personal paypal is brian at freeswitch.org or  
you can use chipin http://www.chipin.com/contribute/id/ddc094318ae5ab5b

I would like to thank Bandwidth.com and Teliax for their support... I  
hope others will help out!

Also we have another sounds order going in sometime near the end of  
January so any money that exceeds the server cost will go to that.   
The sound order I placed monday will be in this Friday.

Happy Holidays and wish you the best for 2009!

Thanks,
Brian West
FreeSWITCH




From jflowers at ezo.net  Tue Dec 16 11:56:46 2008
From: jflowers at ezo.net (jflowers)
Date: Tue, 16 Dec 2008 11:56:46 -0800 (PST)
Subject: [Freeswitch-users] Speed Dial Emulation
In-Reply-To: <154FFEA4-3E17-4570-B8D6-E54ED2D4A028@freeswitch.org>
References: <21016167.post@talk.nabble.com>
	<686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>
	<154FFEA4-3E17-4570-B8D6-E54ED2D4A028@freeswitch.org>
Message-ID: <21040508.post@talk.nabble.com>


What db application (hint hint)?;-)

So maybe what I have to do is "^\d$" and lookup $1 in a db to find the
number to dial?


Brian West-3 wrote:
> 
> Tie that to the db application and you'll have it.
-- 
View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21040508.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Tue Dec 16 12:02:07 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 14:02:07 -0600
Subject: [Freeswitch-users] Speed Dial Emulation
In-Reply-To: <21040508.post@talk.nabble.com>
References: <21016167.post@talk.nabble.com>
	<686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>
	<154FFEA4-3E17-4570-B8D6-E54ED2D4A028@freeswitch.org>
	<21040508.post@talk.nabble.com>
Message-ID: <B0486A14-661C-4D7F-87A4-F80B0585DDB5@freeswitch.org>

Its called "db" :P  check the default dialplan for the insert and  
select examples

/b

On Dec 16, 2008, at 1:56 PM, jflowers wrote:

>
> What db application (hint hint)?;-)
>
> So maybe what I have to do is "^\d$" and lookup $1 in a db to find the
> number to dial?



From jlists at skopis.com  Tue Dec 16 17:50:09 2008
From: jlists at skopis.com (John Skopis (Lists))
Date: Tue, 16 Dec 2008 19:50:09 -0600
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <4947A61F.6060806@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>	<494198F3.10806@ydeasolutions.com.br>
	<4947A61F.6060806@ydeasolutions.com.br>
Message-ID: <49485AD1.5070708@skopis.com>

vinicius wrote:
> hi ppl.. i tried to find something at google, but i couldnt manage to find
> anything.
> i still dont know what to do to make the mod_xml_ldap work.
> i couldnt find information about how to build a config file for the
> module, and where to store it...
> 
> can anyone give me a help?
> 

Be advised mod_xml_ldap is probably not production quality and will
undoubtedly change, eventually at least.

Here is what I used once:

  <bindings>


    <binding name="directory">
		<!--%s is populated with the extension -->
		<param name="filter" value="(FSid=%s)" bindings="directory"/>
		<!--basedn for the searches %s is replaced with domain-->
		<param name="basedn" value="ou=people,dc=example" />
		<param name="url" value="ldap://172.16.75.129" />
		<param name="binddn" value="cn=admin,dc=example" />
		<param name="bindpass" value="secret" />

		<trans>
			<!-- we need to translate these attrs into FS attrs -->
			<tran name="id" mapfrom="FSid" />
			<tran name="mailbox" mapfrom="FSmailbox" />
			<tran name="password" mapfrom="FSPassword" />
			<tran name="vm-password" mapfrom="FSvm-password" />
			<tran name="email-addr" mapfrom="FSemail-addr" />
			<tran name="vm-email-all-messages" mapfrom="FSvm-email-all-messages" />
			<tran name="vm-delete-file" mapfrom="FSvm-delete-file" />
			<tran name="vm-attach-file" mapfrom="FSvm-attach-file" />
		</trans>
    </binding>

    <binding name="configuration">
        <param name="filter" value="(%s=%s)" bindings="configuration"/>
        <param name="basedn" value="name=%s,dc=example" />
        <param name="url" value="ldap://172.16.75.129" />
        <param name="binddn" value="cn=admin,dc=example" />
        <param name="bindpass" value="secret" />
    </binding>
</bindings>


which should/probably/might work with ldap objects like these:

dn: cn=John Skopis,ou=people,dc=example
objectClass: person
objectClass: inetOrgPerson
objectClass: organizationalPerson
objectClass: FreeSWITCH-Exten-Object
objectClass: top
cn: John Skopis
sn: Skopis
givenName: John
FSid: 1001
FSmailbox: 1001
FSpassword: 1234
FSvm-password: 1001
FSemail-addr: john+fs at skopis.com
FSvm-email-all-messages: TRUE
FSvm-delete-file: TRUE
FSvm-attach-file: TRUE

dn: SIPIdentityUserName=1001,ou=h350,dc=example
objectClass: person
objectClass: SIPIdentity
objectClass: top
cn: 1001
sn: 1001
SIPIdentitySIPURI: sip:1001 at 172.16.75.129
SIPIdentityRegistrarAddress: 172.16.75.128
SIPIdentityProxyAddress: 172.16.75.128
SIPIdentityPassword: 1234
SIPIdentityUserName: 1001
SIPIdentityServiceLevel: premium



From marc at kasteris.com  Tue Dec 16 20:24:09 2008
From: marc at kasteris.com (Marc Orenberg)
Date: Tue, 16 Dec 2008 20:24:09 -0800 (PST)
Subject: [Freeswitch-users] Ending a bridged call with a touchtone
Message-ID: <52400.8491.qm@web50803.mail.re2.yahoo.com>

Hello.? I'm trying to allow the A-leg of a bridged call to be able to press a touchtone to end the call.
In my Python script, I set-up a DTMF callback function using setInputCallback, but it doesn't seem to?have any effect?during bridged calls. Is there another way to do this? Thanks.
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From brian at freeswitch.org  Tue Dec 16 09:30:46 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 11:30:46 -0600
Subject: [Freeswitch-users] Ending a bridged call with a touchtone
In-Reply-To: <52400.8491.qm@web50803.mail.re2.yahoo.com>
References: <52400.8491.qm@web50803.mail.re2.yahoo.com>
Message-ID: <F9E12702-5CA0-45A8-B408-3F2162F94A8E@freeswitch.org>

Try bind_meta, examples are in the default dialplan.

/b

On Dec 16, 2008, at 10:24 PM, Marc Orenberg wrote:

> Hello.  I'm trying to allow the A-leg of a bridged call to be able  
> to press a touchtone to end the call.
> In my Python script, I set-up a DTMF callback function using  
> setInputCallback, but it doesn't seem to have any effect during  
> bridged calls. Is there another way to do this? Thanks.

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From jason at jasonjgw.net  Tue Dec 16 21:07:01 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 17 Dec 2008 16:07:01 +1100
Subject: [Freeswitch-users] error creating IPv6 profile when corresponding
	IPv4 port in use
Message-ID: <20081217050701.GA14997@jdc.jasonjgw.net>

Here's the scenario (FreeSWITCH revision 10725).

I have Asterisk listening on port 5060 under IPv4. This version of Asterisk
does not support IPv6, and netstat -6 -a | grep sip suggests that, as
expected, it isn't listening on port 5060 under IPv6.

If I set up FreeSWITCH profiles to listen on port 5070, for example, under
both IPv4 and IPv6 then it works fine.

However, if I create an IPv4 profile to listen on port 5070, and an IPv6
profile to listen on port 5060, Sofia fails with a "port already in use"
error, even though nothing should be using port 5060 under IPv6.

>From the logs:

tport_create(): 0x1b4d0e0
tport_bind_server(0x1b4d0e0) to */[2001:44b8:61::3b]:5060/sip
tport_bind_server(0x1b4d0e0): calling tport_listen for udp
tport_alloc_primary(0x1b4d0e0): new primary tport 0x1b4f010
tport_listen(0x1b4d0e0): bind(pf=10 udp/[2001:44b8:61::3b]:5060): Address already in use
tport_destroy(0x1b4d0e0)
nta: bind([2001:44b8:61::3b]:5060;transport=*): Address already in use
nua: initializing SIP stack failed

Obviously, I can easily use a different port, and hence this issue doesn't
affect me. However, if it's a bug, I think it should still go into the bug
list. I realize the problem might not be in Sofia or FreeSWITCH. Is Asterisk
really taking over the IPv6 port as well?

Environment: Debian Sid, kernel 2.6.26, x86_64 architecture.

Note: once I learn enough about FreeSWITCH to configure it properly and set it
up to meet my personal telephony needs, the Asterisk installation will be,
shall we say, redundant.



From brian at freeswitch.org  Tue Dec 16 21:10:19 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 23:10:19 -0600
Subject: [Freeswitch-users] error creating IPv6 profile when
	corresponding IPv4 port in use
In-Reply-To: <20081217050701.GA14997@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
Message-ID: <4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>

Well the OS reports its in use when we try to bind not much we can do  
about that ... what does netstat -na | grep 5060 show?

/b

On Dec 16, 2008, at 11:07 PM, Jason White wrote:

> nta: bind([2001:44b8:61::3b]:5060;transport=*): Address already in use



From jason at jasonjgw.net  Tue Dec 16 21:19:35 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 17 Dec 2008 16:19:35 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
Message-ID: <20081217051935.GA15357@jdc.jasonjgw.net>

On Tue, Dec 16, 2008 at 11:10:19PM -0600, Brian West wrote:
> Well the OS reports its in use when we try to bind not much we can do  
> about that ... what does netstat -na | grep 5060 show?

udp        0      0 0.0.0.0:5060            0.0.0.0:*

I'll take it, then, that this is a non-issue?



From jason at jasonjgw.net  Tue Dec 16 23:40:02 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 17 Dec 2008 18:40:02 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <20081217051935.GA15357@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
Message-ID: <20081217074002.GA16365@jdc.jasonjgw.net>

The code in bind6only_check in libs/sofia-sip/libsofia-sip-ua/tport/tport.c looks
correct to me, but I can't find where the result is tested (it's in
mr_bindv6only). When bind6only_check() is called in tport_bind_server(), the
return value isn't tested, and I'm having difficulty finding where it is used
- I'm interested in whether we're in fact attempting to bind only to the IPv6
  port and whether the logic is correct here.

When I find the time, I could rebuild with debug symbols and run it under gdb.



From mszlazak at aol.com  Wed Dec 17 00:06:49 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 17 Dec 2008 03:06:49 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
In-Reply-To: <957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com>
References: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>
	<957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com>
Message-ID: <8CB2E358C58A406-9B8-1883@WEBMAIL-MA13.sysops.aol.com>


 Hi Mike,

That does get the audio go between the softphone and the application (Voxeo's Prophecy ASR) "around" FreeSwitch but I would like the audio going "through" FreeSwitch. I plan to do something to it before passing it on.

Support from Voxeo had this to say about the "bypass media" setting and if you could add some more insight that would be much appreciated. Since this is all on one Windows XP machine they can't get the info from the pcap file and are requesting I set up freeswitch on another machine which I will do. I thought you may have some more input.

?

Mark,

This is great news, it certainly confirms our suspicions that freeswitch was not 
forwarding media to Prophecy, or if so, it was doing it on a different port then 
we specified to be listening on.

To address the lingering question in this thread, I don't believe we have a firm 
enough grasp on your deployment calls to understand whether free-switch need the 
RTP stream or not.  If FreeSwitch is intended in your deployment to act as a 
front end for call routing to terminate calls to Prophecy then there is no need 
for it to listen to media.

Of course, it will hold the SIP communication tether so that it remains aware of 
disconnect events, would be my assumption, I am sure freeswitch can verify this 
behavior.

In order for us to understand why this config change is required will need a 
wireshark trace, and with your stacked approach to have both Prophecy and 
freeswitch on the
 same box makes this impossible.  For troubleshooting, if you 
moved freeswitch to another server temporarily, this may offer some insight into 
this problem, with wireshark at our disposal.

Hope this helps!



 


 

-----Original Message-----
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, 16 Dec 2008 5:24 am
Subject: Re: [Freeswitch-users] Help with routing sound locally through FS









If its all local you can also just use:



http://wiki.freeswitch.org/wiki/Bypass_Media




If your still trying to figure it out it could be any number of things, but most relating to misconfigured endpoints or freeswitch, take a look at the sip trace and make sure everything is using the right ip addresses instead of using internal when they should be external or the other way around.




Mike




On Dec 16, 2008, at 5:02 AM, mszlazak at aol.com wrote:


I'm making a call internally from a soft phone to an extension that is suppose to bridge the call internally to another application on the same computer. The applications logs indicate that a connection was made but sound is not being passed back from the application through freeswitch to the softphone. There maybe an issue with rtp timing and associated ports but I'm very new at diagnosing this and fixing the problem.
 
 I've attached both a copy of the FS log and an associated pcap file.
 
 It's all on Windows XP. 
 
 Could someone please take a look.
 
 Thanks.
 




Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 <free.zip>_______________________________________________
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=


 





_______________________________________________
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From carole.olivier at enst.fr  Wed Dec 17 02:06:39 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 02:06:39 -0800 (PST)
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
	<191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
Message-ID: <21050314.post@talk.nabble.com>


Thanks, this works fine.

But I try to use some other API commands and something goes wrong: I would
like to be able to use the API commands for the conference like lock,
unlock, say, etc... from the dialplan.

I try to add in my dialplan 
<action application="bind_meta_app" data="1 a s conference::conf1 lock"/>
but it did not work, I presse F8 and I have got:

2008-12-17 10:47:58 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes Processing
meta digit '2' [conference::conf1 lock]
2008-12-17 10:47:58 [DEBUG] switch_core_session.c:611
switch_core_session_queue_private_event() Kill
sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes [BREAK]

You told me that the API commands should not be blocked isn't it? So I do
not understand where I am wrong. Is "bind_meta_app" not supported in
conference?

I have also tried to write directly:
<action application="conference" data="conf1 lock"/>
which also has no effect.

Thanks for your help,
Carole


Anthony Minessale-2 wrote:
> 
> it should work with
> <action application="bind_meta_app" data="1 ab s transfer::-both 3333"/>
> 
> make sure you have debug log on (press f8) to see if there are any issues.
> 
> 
> On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr> wrote:
> 
>>
>> Thanks for this answer.
>> Just a question so that everything is clear: there is no command to
>> bridge
>> directly a call and both legs into a conference? We have to transfer the
>> call to another extension and from there create the conference isn't?
>>
>> I used the command transfer but I still have a problem. I do the
>> following
>> and it did not work: in my dialplan I write:
>> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
>>
>> The extension 3333 works fine.
>>
>> However, nothing happens when I press *1.
>> Do you have an idea where am I wrong?
>>
>> (A subsidiary question: will both legs be transferred to the extension
>> 3333
>> or just the one which press *1? is there a way to transfer both
>> together?)
>>
>>
>> Thanks a lot,
>> Carole
>>
>>
>> Anthony Minessale-2 wrote:
>> >
>> > There is a transfer dial plan application also.
>> >
>> > There is not really any benefit in blocking the api commands from the
>> > dialplan
>> > apart from the potential for a blocking api call to delay the audio
>> stream
>> > which
>> > you can do at your own risk and use the sleep application with 0ms to
>> > flush
>> > the read buffers.
>> >
>> > So I guess i can lift that limitation in tree.
>> >
>> > see r10790
>> >
>> >
>> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
>> wrote:
>> >
>> >>
>> >> Hello,
>> >>
>> >> I have a general question about the API commands. Some of them are not
>> >> available in the dialplan like uuid_transfer. I was wondering how to
>> call
>> >> an
>> >> API command without using the CLI. Especially I would be interested in
>> >> knowing if there is any way to call them from a phone, I mean bind a
>> key
>> >> to
>> >> an API command.
>> >>
>> >> For instance I would like to transfer both members of a simple call
>> into
>> >> a
>> >> conference by dialing *1.
>> >> I have seen the transfer is possible from the CLI by doing:
>> >> api uuid_transfer <uuid> -both 3001
>> >> but I do not know how to do it else.
>> >>
>> >> If somebody could give me an insight about the topic it would be
>> great.
>> >> Thanks,
>> >> Carole
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> --
>> >> View this message in context:
>> >>
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >>
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com><
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >
>> >
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org><
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> > iax:guest at conference.freeswitch.org/888
>> >
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21050314.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From carole.olivier at enst.fr  Wed Dec 17 02:15:51 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 02:15:51 -0800 (PST)
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <21050314.post@talk.nabble.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
	<191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
	<21050314.post@talk.nabble.com>
Message-ID: <21050436.post@talk.nabble.com>


(I have just read the post again, I have written <action
application="bind_meta_app" data="1 a s conference::conf1 lock"/> but I
meant <action application="bind_meta_app" data="2 a s conference::conf1
lock"/> please don't pay attention for that, I made the mistake when I have
copied it in the post, not in the configuration.)


Carole O. wrote:
> 
> Thanks, this works fine.
> 
> But I try to use some other API commands and something goes wrong: I would
> like to be able to use the API commands for the conference like lock,
> unlock, say, etc... from the dialplan.
> 
> I try to add in my dialplan 
> <action application="bind_meta_app" data="1 a s conference::conf1 lock"/>
> but it did not work, I presse F8 and I have got:
> 
> 2008-12-17 10:47:58 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
> sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes Processing
> meta digit '2' [conference::conf1 lock]
> 2008-12-17 10:47:58 [DEBUG] switch_core_session.c:611
> switch_core_session_queue_private_event() Kill
> sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes [BREAK]
> 
> You told me that the API commands should not be blocked isn't it? So I do
> not understand where I am wrong. Is "bind_meta_app" not supported in
> conference?
> 
> I have also tried to write directly:
> <action application="conference" data="conf1 lock"/>
> which also has no effect.
> 
> Thanks for your help,
> Carole
> 
> 
> Anthony Minessale-2 wrote:
>> 
>> it should work with
>> <action application="bind_meta_app" data="1 ab s transfer::-both 3333"/>
>> 
>> make sure you have debug log on (press f8) to see if there are any
>> issues.
>> 
>> 
>> On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr>
>> wrote:
>> 
>>>
>>> Thanks for this answer.
>>> Just a question so that everything is clear: there is no command to
>>> bridge
>>> directly a call and both legs into a conference? We have to transfer the
>>> call to another extension and from there create the conference isn't?
>>>
>>> I used the command transfer but I still have a problem. I do the
>>> following
>>> and it did not work: in my dialplan I write:
>>> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
>>>
>>> The extension 3333 works fine.
>>>
>>> However, nothing happens when I press *1.
>>> Do you have an idea where am I wrong?
>>>
>>> (A subsidiary question: will both legs be transferred to the extension
>>> 3333
>>> or just the one which press *1? is there a way to transfer both
>>> together?)
>>>
>>>
>>> Thanks a lot,
>>> Carole
>>>
>>>
>>> Anthony Minessale-2 wrote:
>>> >
>>> > There is a transfer dial plan application also.
>>> >
>>> > There is not really any benefit in blocking the api commands from the
>>> > dialplan
>>> > apart from the potential for a blocking api call to delay the audio
>>> stream
>>> > which
>>> > you can do at your own risk and use the sleep application with 0ms to
>>> > flush
>>> > the read buffers.
>>> >
>>> > So I guess i can lift that limitation in tree.
>>> >
>>> > see r10790
>>> >
>>> >
>>> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
>>> wrote:
>>> >
>>> >>
>>> >> Hello,
>>> >>
>>> >> I have a general question about the API commands. Some of them are
>>> not
>>> >> available in the dialplan like uuid_transfer. I was wondering how to
>>> call
>>> >> an
>>> >> API command without using the CLI. Especially I would be interested
>>> in
>>> >> knowing if there is any way to call them from a phone, I mean bind a
>>> key
>>> >> to
>>> >> an API command.
>>> >>
>>> >> For instance I would like to transfer both members of a simple call
>>> into
>>> >> a
>>> >> conference by dialing *1.
>>> >> I have seen the transfer is possible from the CLI by doing:
>>> >> api uuid_transfer <uuid> -both 3001
>>> >> but I do not know how to do it else.
>>> >>
>>> >> If somebody could give me an insight about the topic it would be
>>> great.
>>> >> Thanks,
>>> >> Carole
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> View this message in context:
>>> >>
>>> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> Freeswitch-users mailing list
>>> >> Freeswitch-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> <MSN%3Aanthony_minessale at hotmail.com><
>>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>>> >
>>> >
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>>> >
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> <sip%3A888 at conference.freeswitch.org><
>>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>>> >
>>> > iax:guest at conference.freeswitch.org/888
>>> >
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>>> >
>>> > pstn:213-799-1400
>>> >
>>> > _______________________________________________
>>> > Freeswitch-users mailing list
>>> > Freeswitch-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 

-- 
View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21050436.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From fidibus83 at aol.com  Wed Dec 17 02:17:41 2008
From: fidibus83 at aol.com (fidibus83)
Date: Wed, 17 Dec 2008 11:17:41 +0100
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <B4F10400-D789-4507-8C06-F2B7F79ECED7@freeswitch.org>
References: <002101c95f64$56958e60$6445310a@Franzi><7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com><007601c95f85$c1462aa0$6445310a@Franzi><9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org><009f01c95f8d$7918ddb0$6445310a@Franzi>
	<B4F10400-D789-4507-8C06-F2B7F79ECED7@freeswitch.org>
Message-ID: <004201c96030$b2b9fd80$6445310a@Franzi>

I did a reinstall but there is the same error!

Is there something else I can do to remove the error?

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael S Collins
Gesendet: Dienstag, 16. Dezember 2008 16:53
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

Thanks for trying. You might want to clean out your zaptel install and do a
complete reinstall. 

 

-MC

Sent from my iPhone


On Dec 16, 2008, at 6:49 AM, "fidibus83" <fidibus83 at aol.com> wrote:

It?s already the same error.

 


  _____  


Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael S Collins
Gesendet: Dienstag, 16. Dezember 2008 15:37
An:  <mailto:freeswitch-users at lists.freeswitch.org>
freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

Just a hunch but try removing the spaces in this line:

 

loadzone=de

Zaptel can be quirky. 

-MC


Sent from my iPhone


On Dec 16, 2008, at 5:54 AM, "fidibus83" < <mailto:fidibus83 at aol.com>
fidibus83 at aol.com> wrote:

I have installed zaptel-1.4.11 

I have looked in zonedata.c and there is configured de-tonezone 

 


  _____  


Von:  <mailto:freeswitch-users-bounces at lists.freeswitch.org>
freeswitch-users-bounces at lists.freeswitch.org [
<mailto:freeswitch-users-bounces at lists.freeswitch.org>
mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael
Jerris
Gesendet: Dienstag, 16. Dezember 2008 14:20
An:  <mailto:freeswitch-users at lists.freeswitch.org>
<mailto:freeswitch-users at lists.freeswitch.org>
freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

It sounds like there is no de tonezone in the zaptel drivers, but I can't
imagine thats true.  What version of the drivers do you have installed?  

 

 

On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:









Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

 

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg ?vv:

 

31 channels configured.

 

ioctl(ZT_LOADZONE) failed: Invalid argument

Notice: Configuration file is /etc/zaptel.conf

line 288: Unable to register tone zone 'de'

 

 

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

What do I wrong? Please help me!

_______________________________________________
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 <mailto:Freeswitch-users at lists.freeswitch.org>
<mailto:Freeswitch-users at lists.freeswitch.org>
Freeswitch-users at lists.freeswitch.org
 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
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UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>
<http://lists.freeswitch.org/mailman/options/freeswitch-users>
http://lists.freeswitch.org/mailman/options/freeswitch-users
 <http://www.freeswitch.org>  <http://www.freeswitch.org>
http://www.freeswitch.org

 

= 

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 <mailto:Freeswitch-users at lists.freeswitch.org>
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= h.org
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= y> = 

_______________________________________________
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From carole.olivier at enst.fr  Wed Dec 17 05:34:08 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 05:34:08 -0800 (PST)
Subject: [Freeswitch-users]  dynamic conference
Message-ID: <21053181.post@talk.nabble.com>


Hello,

I have done a small change in my dialplan which works but since I am new
with FreeSWITCH I was wondering if this solution goes with the philosophy of
the software or if it is absurd and there is a solution more adapted .

I try to reproduce the following functionality:
"A and B are on a simple call and decide to add C and have a conference.
Later on they decide also to invite D..."

In the dialplan I have added (based on the default dialplan):
<extension name="Local_Extension">
     <condition field="destination_number" expression="^(10[01][0-9])$">
           ................
          <anti-action application="bind_meta_app" data="1 a o transfer::
transf_call_${caller_id_number}"/>
          <anti-action application="bind_meta_app" data="2 a s
transfer::-both transf_call_${caller_id_number}"/>
          .......
      </condition>
</extension>

<extension name="transf_call_to_conf">
      <condition field="destination_number" expression="^transf_call_(.*)$">
         <action application="answer"/>
         <action application="conference" data="conf$1 at conf_dynamic"/>
       </condition>
</extension>

Here, if A calls B then A can bridge both legs into a conference named
"confnumberofA" by dialing *2.
Then, A can put this call on hold and call C. 
- If C answers and agrees then A can press *1 in order to bridge C into the
same conference "confnumberofA". A will come back into the conference which
is still on hold. 
- If C does not answer then A will still be able to come back into the
conference it puts on hold.

My main problem is the name of the conference. Since everybody should be
able to convert a simple call into a conference, the conference's name has
to be unique each time. I have chosen to make it dependent on the caller
number which is not perfect because then he is the only one which can bridge
the call and add member. However, I do not have any other idea, maybe I have
missed another possibility.

I would be glad to get any critics you have, it would help me to better
understand the fundamental concepts.

Thanks,
Carole
-- 
View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21053181.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From vkobashi at ydeasolutions.com.br  Wed Dec 17 05:45:03 2008
From: vkobashi at ydeasolutions.com.br (vinicius)
Date: Wed, 17 Dec 2008 11:45:03 -0200
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49485AD1.5070708@skopis.com>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>	<494198F3.10806@ydeasolutions.com.br>	<4947A61F.6060806@ydeasolutions.com.br>
	<49485AD1.5070708@skopis.com>
Message-ID: <4949025F.9040008@ydeasolutions.com.br>

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From freeswitch-users at lists.rupa.com  Wed Dec 17 05:47:45 2008
From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists))
Date: Wed, 17 Dec 2008 07:47:45 -0600
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21053181.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com>
Message-ID: <49490301.8040701@lists.rupa.com>

On 12/17/2008 7:34 AM, Carole O. wrote:
> My main problem is the name of the conference. Since everybody should be
> able to convert a simple call into a conference, the conference's name has
> to be unique each time. I have chosen to make it dependent on the caller
> number which is not perfect because then he is the only one which can bridge
> the call and add member. However, I do not have any other idea, maybe I have
> missed another possibility.

If the conf name has to be unique, why not ensure that by making the
conference name based on the uuid of the a-leg?

> Thanks,
> Carole



From anthony.minessale at gmail.com  Wed Dec 17 06:19:31 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Dec 2008 08:19:31 -0600
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <21050436.post@talk.nabble.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
	<191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
	<21050314.post@talk.nabble.com> <21050436.post@talk.nabble.com>
Message-ID: <191c3a030812170619l27c9a7b3n55321efa1fa80f60@mail.gmail.com>

I said i unblocked the ones in mod_commands

mod_conference was it's own module.
I changed it to work in latest trunk as well.



On Wed, Dec 17, 2008 at 4:15 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> (I have just read the post again, I have written <action
> application="bind_meta_app" data="1 a s conference::conf1 lock"/> but I
> meant <action application="bind_meta_app" data="2 a s conference::conf1
> lock"/> please don't pay attention for that, I made the mistake when I have
> copied it in the post, not in the configuration.)
>
>
> Carole O. wrote:
> >
> > Thanks, this works fine.
> >
> > But I try to use some other API commands and something goes wrong: I
> would
> > like to be able to use the API commands for the conference like lock,
> > unlock, say, etc... from the dialplan.
> >
> > I try to add in my dialplan
> > <action application="bind_meta_app" data="1 a s conference::conf1 lock"/>
> > but it did not work, I presse F8 and I have got:
> >
> > 2008-12-17 10:47:58 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
> > sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes
> Processing
> > meta digit '2' [conference::conf1 lock]
> > 2008-12-17 10:47:58 [DEBUG] switch_core_session.c:611
> > switch_core_session_queue_private_event() Kill
> > sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes [BREAK]
> >
> > You told me that the API commands should not be blocked isn't it? So I do
> > not understand where I am wrong. Is "bind_meta_app" not supported in
> > conference?
> >
> > I have also tried to write directly:
> > <action application="conference" data="conf1 lock"/>
> > which also has no effect.
> >
> > Thanks for your help,
> > Carole
> >
> >
> > Anthony Minessale-2 wrote:
> >>
> >> it should work with
> >> <action application="bind_meta_app" data="1 ab s transfer::-both 3333"/>
> >>
> >> make sure you have debug log on (press f8) to see if there are any
> >> issues.
> >>
> >>
> >> On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr>
> >> wrote:
> >>
> >>>
> >>> Thanks for this answer.
> >>> Just a question so that everything is clear: there is no command to
> >>> bridge
> >>> directly a call and both legs into a conference? We have to transfer
> the
> >>> call to another extension and from there create the conference isn't?
> >>>
> >>> I used the command transfer but I still have a problem. I do the
> >>> following
> >>> and it did not work: in my dialplan I write:
> >>> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
> >>>
> >>> The extension 3333 works fine.
> >>>
> >>> However, nothing happens when I press *1.
> >>> Do you have an idea where am I wrong?
> >>>
> >>> (A subsidiary question: will both legs be transferred to the extension
> >>> 3333
> >>> or just the one which press *1? is there a way to transfer both
> >>> together?)
> >>>
> >>>
> >>> Thanks a lot,
> >>> Carole
> >>>
> >>>
> >>> Anthony Minessale-2 wrote:
> >>> >
> >>> > There is a transfer dial plan application also.
> >>> >
> >>> > There is not really any benefit in blocking the api commands from the
> >>> > dialplan
> >>> > apart from the potential for a blocking api call to delay the audio
> >>> stream
> >>> > which
> >>> > you can do at your own risk and use the sleep application with 0ms to
> >>> > flush
> >>> > the read buffers.
> >>> >
> >>> > So I guess i can lift that limitation in tree.
> >>> >
> >>> > see r10790
> >>> >
> >>> >
> >>> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
> >>> wrote:
> >>> >
> >>> >>
> >>> >> Hello,
> >>> >>
> >>> >> I have a general question about the API commands. Some of them are
> >>> not
> >>> >> available in the dialplan like uuid_transfer. I was wondering how to
> >>> call
> >>> >> an
> >>> >> API command without using the CLI. Especially I would be interested
> >>> in
> >>> >> knowing if there is any way to call them from a phone, I mean bind a
> >>> key
> >>> >> to
> >>> >> an API command.
> >>> >>
> >>> >> For instance I would like to transfer both members of a simple call
> >>> into
> >>> >> a
> >>> >> conference by dialing *1.
> >>> >> I have seen the transfer is possible from the CLI by doing:
> >>> >> api uuid_transfer <uuid> -both 3001
> >>> >> but I do not know how to do it else.
> >>> >>
> >>> >> If somebody could give me an insight about the topic it would be
> >>> great.
> >>> >> Thanks,
> >>> >> Carole
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >> --
> >>> >> View this message in context:
> >>> >>
> >>>
> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
> >>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>> >>
> >>> >>
> >>> >> _______________________________________________
> >>> >> Freeswitch-users mailing list
> >>> >> Freeswitch-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> UNSUBSCRIBE:
> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >>
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Anthony Minessale II
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> >
> >>> > AIM: anthm
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> <MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> ><
> >>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> <MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >
> >>> >
> >>> >
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> <PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >
> >>> >
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> <sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> ><
> >>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> <sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >
> >>> >
> >>> > iax:guest at conference.freeswitch.org/888
> >>> >
> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> <googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >
> >>> >
> >>> > pstn:213-799-1400
> >>> >
> >>> > _______________________________________________
> >>> > Freeswitch-users mailing list
> >>> > Freeswitch-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> --
> >>> View this message in context:
> >>>
> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>>
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >> pstn:213-799-1400
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/general-question-about-API-command-tp21032754p21050436.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From carole.olivier at enst.fr  Wed Dec 17 06:24:28 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 06:24:28 -0800 (PST)
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <49490301.8040701@lists.rupa.com>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
Message-ID: <21054080.post@talk.nabble.com>


It would be unique you are right but I am not sure I can get its value if A
puts the call on hold, calls C and wants to add it to the conference whose
name dependent of the uuid of another session. 
I think if I use ${uuid} to add C I will have the uuid of the session
between A and C and not A and B no?
And I really have to configure this from the dialplan so statically.

Am I wrong somewhere??

Carole


Rupa Schomaker (lists)-2 wrote:
> 
> On 12/17/2008 7:34 AM, Carole O. wrote:
>> My main problem is the name of the conference. Since everybody should be
>> able to convert a simple call into a conference, the conference's name
>> has
>> to be unique each time. I have chosen to make it dependent on the caller
>> number which is not perfect because then he is the only one which can
>> bridge
>> the call and add member. However, I do not have any other idea, maybe I
>> have
>> missed another possibility.
> 
> If the conf name has to be unique, why not ensure that by making the
> conference name based on the uuid of the a-leg?
> 
>> Thanks,
>> Carole
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21054080.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From carole.olivier at enst.fr  Wed Dec 17 06:31:48 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 06:31:48 -0800 (PST)
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <191c3a030812170619l27c9a7b3n55321efa1fa80f60@mail.gmail.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
	<191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
	<21050314.post@talk.nabble.com> <21050436.post@talk.nabble.com>
	<191c3a030812170619l27c9a7b3n55321efa1fa80f60@mail.gmail.com>
Message-ID: <21054214.post@talk.nabble.com>


ok 
Thanks a lot,
Carole


Anthony Minessale-2 wrote:
> 
> I said i unblocked the ones in mod_commands
> 
> mod_conference was it's own module.
> I changed it to work in latest trunk as well.
> 
> 
> 
> On Wed, Dec 17, 2008 at 4:15 AM, Carole O. <carole.olivier at enst.fr> wrote:
> 
>>
>> (I have just read the post again, I have written <action
>> application="bind_meta_app" data="1 a s conference::conf1 lock"/> but I
>> meant <action application="bind_meta_app" data="2 a s conference::conf1
>> lock"/> please don't pay attention for that, I made the mistake when I
>> have
>> copied it in the post, not in the configuration.)
>>
>>
>> Carole O. wrote:
>> >
>> > Thanks, this works fine.
>> >
>> > But I try to use some other API commands and something goes wrong: I
>> would
>> > like to be able to use the API commands for the conference like lock,
>> > unlock, say, etc... from the dialplan.
>> >
>> > I try to add in my dialplan
>> > <action application="bind_meta_app" data="1 a s conference::conf1
>> lock"/>
>> > but it did not work, I presse F8 and I have got:
>> >
>> > 2008-12-17 10:47:58 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
>> > sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes
>> Processing
>> > meta digit '2' [conference::conf1 lock]
>> > 2008-12-17 10:47:58 [DEBUG] switch_core_session.c:611
>> > switch_core_session_queue_private_event() Kill
>> > sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes [BREAK]
>> >
>> > You told me that the API commands should not be blocked isn't it? So I
>> do
>> > not understand where I am wrong. Is "bind_meta_app" not supported in
>> > conference?
>> >
>> > I have also tried to write directly:
>> > <action application="conference" data="conf1 lock"/>
>> > which also has no effect.
>> >
>> > Thanks for your help,
>> > Carole
>> >
>> >
>> > Anthony Minessale-2 wrote:
>> >>
>> >> it should work with
>> >> <action application="bind_meta_app" data="1 ab s transfer::-both
>> 3333"/>
>> >>
>> >> make sure you have debug log on (press f8) to see if there are any
>> >> issues.
>> >>
>> >>
>> >> On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr>
>> >> wrote:
>> >>
>> >>>
>> >>> Thanks for this answer.
>> >>> Just a question so that everything is clear: there is no command to
>> >>> bridge
>> >>> directly a call and both legs into a conference? We have to transfer
>> the
>> >>> call to another extension and from there create the conference isn't?
>> >>>
>> >>> I used the command transfer but I still have a problem. I do the
>> >>> following
>> >>> and it did not work: in my dialplan I write:
>> >>> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
>> >>>
>> >>> The extension 3333 works fine.
>> >>>
>> >>> However, nothing happens when I press *1.
>> >>> Do you have an idea where am I wrong?
>> >>>
>> >>> (A subsidiary question: will both legs be transferred to the
>> extension
>> >>> 3333
>> >>> or just the one which press *1? is there a way to transfer both
>> >>> together?)
>> >>>
>> >>>
>> >>> Thanks a lot,
>> >>> Carole
>> >>>
>> >>>
>> >>> Anthony Minessale-2 wrote:
>> >>> >
>> >>> > There is a transfer dial plan application also.
>> >>> >
>> >>> > There is not really any benefit in blocking the api commands from
>> the
>> >>> > dialplan
>> >>> > apart from the potential for a blocking api call to delay the audio
>> >>> stream
>> >>> > which
>> >>> > you can do at your own risk and use the sleep application with 0ms
>> to
>> >>> > flush
>> >>> > the read buffers.
>> >>> >
>> >>> > So I guess i can lift that limitation in tree.
>> >>> >
>> >>> > see r10790
>> >>> >
>> >>> >
>> >>> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
>> >>> wrote:
>> >>> >
>> >>> >>
>> >>> >> Hello,
>> >>> >>
>> >>> >> I have a general question about the API commands. Some of them are
>> >>> not
>> >>> >> available in the dialplan like uuid_transfer. I was wondering how
>> to
>> >>> call
>> >>> >> an
>> >>> >> API command without using the CLI. Especially I would be
>> interested
>> >>> in
>> >>> >> knowing if there is any way to call them from a phone, I mean bind
>> a
>> >>> key
>> >>> >> to
>> >>> >> an API command.
>> >>> >>
>> >>> >> For instance I would like to transfer both members of a simple
>> call
>> >>> into
>> >>> >> a
>> >>> >> conference by dialing *1.
>> >>> >> I have seen the transfer is possible from the CLI by doing:
>> >>> >> api uuid_transfer <uuid> -both 3001
>> >>> >> but I do not know how to do it else.
>> >>> >>
>> >>> >> If somebody could give me an insight about the topic it would be
>> >>> great.
>> >>> >> Thanks,
>> >>> >> Carole
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >> --
>> >>> >> View this message in context:
>> >>> >>
>> >>>
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
>> >>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >>> >>
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> Freeswitch-users mailing list
>> >>> >> Freeswitch-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> UNSUBSCRIBE:
>> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >>
>> >>> >
>> >>> >
>> >>> >
>> >>> > --
>> >>> > Anthony Minessale II
>> >>> >
>> >>> > FreeSWITCH http://www.freeswitch.org/
>> >>> > ClueCon http://www.cluecon.com/
>> >>> >
>> >>> > AIM: anthm
>> >>> >
>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>>
>> <MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> ><
>> >>>
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> <MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
>> >
>> >>> >
>> >>> >
>> >>>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> >>>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> <PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
>> >
>> >>> >
>> >>> > IRC: irc.freenode.net #freeswitch
>> >>> >
>> >>> > FreeSWITCH Developer Conference
>> >>> >
>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>>
>> <sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> ><
>> >>>
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> <sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
>> >
>> >>> >
>> >>> > iax:guest at conference.freeswitch.org/888
>> >>> >
>> >>>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> >>>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> <googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
>> >
>> >>> >
>> >>> > pstn:213-799-1400
>> >>> >
>> >>> > _______________________________________________
>> >>> > Freeswitch-users mailing list
>> >>> > Freeswitch-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> --
>> >>> View this message in context:
>> >>>
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
>> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> Freeswitch-users mailing list
>> >>> Freeswitch-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com><
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >
>> >>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org><
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> >> iax:guest at conference.freeswitch.org/888
>> >>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> >> pstn:213-799-1400
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21050436.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21054214.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From cstomi.levlist at gmail.com  Wed Dec 17 06:49:34 2008
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Wed, 17 Dec 2008 15:49:34 +0100
Subject: [Freeswitch-users] DNS RV faiover
Message-ID: <4949117E.5090202@gmail.com>

Hello,

We'd like to use DNS SRV for failover.

if we are bridge sofia/profile/whatever at domain.with.srv it works perfectly
but with gateway wich has this record in its proxy parameter it doesn't 
work.
Once we set up an A record too it works, so we assume dialing gateway 
doesn't use SRV records.
Is there any differences between the 2 method?

Thanks any help,
Tamas





From helmut.kuper at ewetel.de  Wed Dec 17 06:53:38 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 17 Dec 2008 15:53:38 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <4947EBB2.7030401@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>	<4947CBCB.8060204@ewetel.de>	<4947DAE1.1050706@ewetel.de>	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
	<4947EBB2.7030401@ewetel.de>
Message-ID: <49491272.9000103@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

I updated the jira bug. I did a Q931/Q921 trace. Currently there is no
direct hint, that FS is doing something wrong. NT side is just not
anwering the SETUP of FS, buuut, I was asked if FS is able to allow NT
side the channel management, so NT says what channel is to use instead
of FS.


Can FS do this? If so, how can I configure that?

Analyzing the trace throws a new bug: Sometimes our EWSD finds a
checksum errors in FS's RR messages.

more here: http://jira.freeswitch.org/browse/OPENZAP-40


If one of FS board wants the whole decoded trace just raise your hand.

regards
Helmut
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From anthony.minessale at gmail.com  Wed Dec 17 07:05:46 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Dec 2008 09:05:46 -0600
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21054080.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
	<21054080.post@talk.nabble.com>
Message-ID: <191c3a030812170705n686bb20buc2bbd002c36e4a49@mail.gmail.com>

you could make up a uuid just for the conference name in the original call

<action application="export" data="my_conf_id=${create_uuid()}"/>

Now this channel and any other channel created by this channel will inherit
this var


On Wed, Dec 17, 2008 at 8:24 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> It would be unique you are right but I am not sure I can get its value if A
> puts the call on hold, calls C and wants to add it to the conference whose
> name dependent of the uuid of another session.
> I think if I use ${uuid} to add C I will have the uuid of the session
> between A and C and not A and B no?
> And I really have to configure this from the dialplan so statically.
>
> Am I wrong somewhere??
>
> Carole
>
>
> Rupa Schomaker (lists)-2 wrote:
> >
> > On 12/17/2008 7:34 AM, Carole O. wrote:
> >> My main problem is the name of the conference. Since everybody should be
> >> able to convert a simple call into a conference, the conference's name
> >> has
> >> to be unique each time. I have chosen to make it dependent on the caller
> >> number which is not perfect because then he is the only one which can
> >> bridge
> >> the call and add member. However, I do not have any other idea, maybe I
> >> have
> >> missed another possibility.
> >
> > If the conf name has to be unique, why not ensure that by making the
> > conference name based on the uuid of the a-leg?
> >
> >> Thanks,
> >> Carole
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/dynamic-conference-tp21053181p21054080.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Dec 17 07:10:49 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Dec 2008 09:10:49 -0600
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <49491272.9000103@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
	<4947EBB2.7030401@ewetel.de> <49491272.9000103@ewetel.de>
Message-ID: <191c3a030812170710u1f85fe97s7121cda8df02a50@mail.gmail.com>

try

<param name="suggest-channel" value="true"/>

in the <span> in openzap.conf.xml


On Wed, Dec 17, 2008 at 8:53 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi,
>
> I updated the jira bug. I did a Q931/Q921 trace. Currently there is no
> direct hint, that FS is doing something wrong. NT side is just not
> anwering the SETUP of FS, buuut, I was asked if FS is able to allow NT
> side the channel management, so NT says what channel is to use instead
> of FS.
>
>
> Can FS do this? If so, how can I configure that?
>
> Analyzing the trace throws a new bug: Sometimes our EWSD finds a
> checksum errors in FS's RR messages.
>
> more here: http://jira.freeswitch.org/browse/OPENZAP-40
>
>
> If one of FS board wants the whole decoded trace just raise your hand.
>
> regards
> Helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklJEnIACgkQ4tZeNddg3dxIdwCbBbTw7a3bG/mnVvVmDSpbH5Bw
> SU0AniHPV4qTIpuI8ENclxXyOn6pcueR
> =Ey+M
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Dec 17 07:13:20 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Dec 2008 09:13:20 -0600
Subject: [Freeswitch-users] DNS RV faiover
In-Reply-To: <4949117E.5090202@gmail.com>
References: <4949117E.5090202@gmail.com>
Message-ID: <191c3a030812170713n31330e75o94b8c2ded1010973@mail.gmail.com>

it might not.

try putting the value in register-proxy as well
sip:host.tld


On Wed, Dec 17, 2008 at 8:49 AM, Tamas Cseke <cstomi.levlist at gmail.com>wrote:

> Hello,
>
> We'd like to use DNS SRV for failover.
>
> if we are bridge sofia/profile/whatever at domain.with.srv it works perfectly
> but with gateway wich has this record in its proxy parameter it doesn't
> work.
> Once we set up an A record too it works, so we assume dialing gateway
> doesn't use SRV records.
> Is there any differences between the 2 method?
>
> Thanks any help,
> Tamas
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From freeswitch-users at lists.rupa.com  Wed Dec 17 07:18:31 2008
From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists))
Date: Wed, 17 Dec 2008 09:18:31 -0600
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21054080.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
	<21054080.post@talk.nabble.com>
Message-ID: <49491847.3030403@lists.rupa.com>

On 12/17/2008 8:24 AM, Carole O. wrote:
> It would be unique you are right but I am not sure I can get its value if A
> puts the call on hold, calls C and wants to add it to the conference whose
> name dependent of the uuid of another session. 
> I think if I use ${uuid} to add C I will have the uuid of the session
> between A and C and not A and B no?
> And I really have to configure this from the dialplan so statically.
> 
> Am I wrong somewhere??
> 
> Carole

Ah, yeah.  uuid would not be the same when initiating a new call that
you then transfer to the conference call.  You need something that is
intrinsic to the endpoint.

I did a quick info dump to an originated call.  Depending on your
use-case (are these calls originating from registered handsets, trunked
from a sip provider, etc) you might want to rely on the variable
"sip_contact_uri" which is a combination of registered user name and ip
(and port if port isn't 5060).  This should be unique per endpoint.




From helmut.kuper at ewetel.de  Wed Dec 17 07:30:22 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 17 Dec 2008 16:30:22 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <191c3a030812170710u1f85fe97s7121cda8df02a50@mail.gmail.com>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>	<4947CBCB.8060204@ewetel.de>
	<4947DAE1.1050706@ewetel.de>	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>	<4947EBB2.7030401@ewetel.de>
	<49491272.9000103@ewetel.de>
	<191c3a030812170710u1f85fe97s7121cda8df02a50@mail.gmail.com>
Message-ID: <49491B0E.8080505@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

thx, but that doesn't work very good. Outgoing calls ring only once and
then this error rises in console:

2008-12-17 16:23:19 [DEBUG] Span:0 Q.931() Sending message to Layer4
(size: 103)
2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got
an event! Type:[02] Size:[103] CRV: 7 (0x7, CTX: Terminator)
2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan
810f590 (1:1) source isdn_data->channels_local_crv[0x7]
2008-12-17 16:23:19 [CRIT] ozmod_isdn.c:701 zap_isdn_931_34() Received
CALL PROCEEDING message for channel 0
2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:702 zap_isdn_931_34() Changing
state on 1:1 from DIALING to PROGRESS

regards
Helmut
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SB0AoIQp+ixmWUGBKyMFXIZQ6AbQsWK0
=I2L3
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From cstomi.levlist at gmail.com  Wed Dec 17 08:05:37 2008
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Wed, 17 Dec 2008 17:05:37 +0100
Subject: [Freeswitch-users] DNS RV faiover
In-Reply-To: <191c3a030812170713n31330e75o94b8c2ded1010973@mail.gmail.com>
References: <4949117E.5090202@gmail.com>
	<191c3a030812170713n31330e75o94b8c2ded1010973@mail.gmail.com>
Message-ID: <49492351.4070101@gmail.com>

Helo,

with register-proxy registrations use SRV.
but it I use register=false param, and dial the gw it lookup only A.

I figured out if I don't specify the port in proxy param the SRV lookup 
is working,
but if I put ":5060" it doen't work. So it is a problem if I don't want 
to use the default port.

Regards,
Tamas

Anthony Minessale ?rta:
> it might not.
>
> try putting the value in register-proxy as well
> sip:host.tld
>
>
> On Wed, Dec 17, 2008 at 8:49 AM, Tamas Cseke <cstomi.levlist at gmail.com>wrote:
>
>   
>> Hello,
>>
>> We'd like to use DNS SRV for failover.
>>
>> if we are bridge sofia/profile/whatever at domain.with.srv it works perfectly
>> but with gateway wich has this record in its proxy parameter it doesn't
>> work.
>> Once we set up an A record too it works, so we assume dialing gateway
>> doesn't use SRV records.
>> Is there any differences between the 2 method?
>>
>> Thanks any help,
>> Tamas
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Wed Dec 17 08:12:22 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Dec 2008 10:12:22 -0600
Subject: [Freeswitch-users] DNS RV faiover
In-Reply-To: <49492351.4070101@gmail.com>
References: <4949117E.5090202@gmail.com>
	<191c3a030812170713n31330e75o94b8c2ded1010973@mail.gmail.com>
	<49492351.4070101@gmail.com>
Message-ID: <040BE086-EDF5-41C3-A805-C651CC7ED126@freeswitch.org>


On Dec 17, 2008, at 10:05 AM, Tamas Cseke wrote:

> Helo,
>
> with register-proxy registrations use SRV.
> but it I use register=false param, and dial the gw it lookup only A.
>
> I figured out if I don't specify the port in proxy param the SRV  
> lookup
> is working,
> but if I put ":5060" it doen't work. So it is a problem if I don't  
> want
> to use the default port.

Thats exactly what it should do you told it the port and address so  
its not going to lookup any SRV records.

/b



>
>
> Regards,
> Tamas



From kristjan.ugrin at gmail.com  Wed Dec 17 06:46:16 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Wed, 17 Dec 2008 15:46:16 +0100
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
Message-ID: <op.uma9zdmxatcbto@linux.site>

I have an idea which is takes too many characters for irc.
I'm relatively new to telephony and such stuff, I managed to get  
freeswitch running, but I don't fully uderstand
my problem in detail and how to solve it, so I need a bit of directions.

Briefly, my idea is to have a jabber contact, which would gtalk user add  
as their buddy. Let's call it callbuddy at somejabbersrv.com.
This would be a bot that would accept commands - I've already made a small  
xmpp java bot which just logs into gtalk and send echo
messages to users - like a parrot, nothing serious. Also I've set up  
openfire jabber server.
What I would like him to do, is when user would tipe "call 0189432443" it  
would initiate a call between contact who tiped in this number (command)  
and this number.

Is this possible, and what would be the best implementation (dingaling  
acting as client...)?

Thanks.

-- 
kriko


From freeswitch at davidnicol.otherinbox.com  Wed Dec 17 07:41:46 2008
From: freeswitch at davidnicol.otherinbox.com (freeswitch at davidnicol.otherinbox.com)
Date: Wed, 17 Dec 2008 10:41:46 -0500
Subject: [Freeswitch-users] Cisco contest
Message-ID: <200812171541.mBHFfj4Y010761@box7.911domain.com>

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From mike at jerris.com  Wed Dec 17 09:10:18 2008
From: mike at jerris.com (Michael Jerris)
Date: Wed, 17 Dec 2008 12:10:18 -0500
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <20081217074002.GA16365@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
Message-ID: <5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>


On Dec 17, 2008, at 2:40 AM, Jason White wrote:

> The code in bind6only_check in libs/sofia-sip/libsofia-sip-ua/tport/ 
> tport.c looks
> correct to me, but I can't find where the result is tested (it's in
> mr_bindv6only). When bind6only_check() is called in  
> tport_bind_server(), the
> return value isn't tested, and I'm having difficulty finding where  
> it is used
> - I'm interested in whether we're in fact attempting to bind only to  
> the IPv6
>  port and whether the logic is correct here.
>
> When I find the time, I could rebuild with debug symbols and run it  
> under gdb.

If this is in fact a bug, could you please report it to the sofia-sip  
bugtracker.  Patches are very helpful there.

Mike



From mike at jerris.com  Wed Dec 17 09:12:42 2008
From: mike at jerris.com (Michael Jerris)
Date: Wed, 17 Dec 2008 12:12:42 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
In-Reply-To: <8CB2E358C58A406-9B8-1883@WEBMAIL-MA13.sysops.aol.com>
References: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>
	<957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com>
	<8CB2E358C58A406-9B8-1883@WEBMAIL-MA13.sysops.aol.com>
Message-ID: <674E25FB-0584-4357-B556-81BF79F84209@jerris.com>

I think the best way to confirm all this is to load a full pcap in  
wireshark and have it pull the wav file of the individual audio  
streams to see what is going on.

Mike

On Dec 17, 2008, at 3:06 AM, mszlazak at aol.com wrote:

> Hi Mike,
>
> That does get the audio go between the softphone and the application  
> (Voxeo's Prophecy ASR) "around" FreeSwitch but I would like the  
> audio going "through" FreeSwitch. I plan to do something to it  
> before passing it on.
>
> Support from Voxeo had this to say about the "bypass media" setting  
> and if you could add some more insight that would be much  
> appreciated. Since this is all on one Windows XP machine they can't  
> get the info from the pcap file and are requesting I set up  
> freeswitch on another machine which I will do. I thought you may  
> have some more input.

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From mike at jerris.com  Wed Dec 17 09:15:20 2008
From: mike at jerris.com (Michael Jerris)
Date: Wed, 17 Dec 2008 12:15:20 -0500
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <004201c96030$b2b9fd80$6445310a@Franzi>
References: <002101c95f64$56958e60$6445310a@Franzi><7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com><007601c95f85$c1462aa0$6445310a@Franzi><9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org><009f01c95f8d$7918ddb0$6445310a@Franzi>
	<B4F10400-D789-4507-8C06-F2B7F79ECED7@freeswitch.org>
	<004201c96030$b2b9fd80$6445310a@Franzi>
Message-ID: <C3A43C47-73FA-4F63-B8D6-1BB053B9207F@jerris.com>

FreeSWITCH/openzap is completely uninvolved at this point, you might  
try asking on the zaptel mailing lists?

Mike

On Dec 17, 2008, at 5:17 AM, fidibus83 wrote:

> I did a reinstall but there is the same error!
> Is there something else I can do to remove the error?
>
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From kirk.bateman at gmail.com  Wed Dec 17 09:15:11 2008
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Wed, 17 Dec 2008 17:15:11 +0000
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
Message-ID: <2bee4fc40812170915p1b5d91feu5fcfbee6713fad40@mail.gmail.com>

Kriko,

I have been looking at the same sort of thing, but I'm planning to implement
an ejabberd bot component (so I can hopefully use the new mod_erlang_event
freeswitch interface).

It seems to me that bits of the current dingaling / jingle interface are
having problems, like not liking sending messages to other domains, its
generally working if they are all .gmail.com users but when you have some on
googlemail.com etc it starts breaking, and doesn't use the whole JID in the
from attribute for sending messages.

When I get a chance I'll try and narrow down the problem.

Cheers

Kirk

Date: Wed, 17 Dec 2008 15:46:16 +0100
> From: kriko <kristjan.ugrin at gmail.com>
> Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
> To: "freeswitch-users at lists.freeswitch.org"
>        <freeswitch-users at lists.freeswitch.org>
> Message-ID: <op.uma9zdmxatcbto at linux.site>
> Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8
>
> I have an idea which is takes too many characters for irc.
> I'm relatively new to telephony and such stuff, I managed to get
> freeswitch running, but I don't fully uderstand
> my problem in detail and how to solve it, so I need a bit of directions.
>
> Briefly, my idea is to have a jabber contact, which would gtalk user add
> as their buddy. Let's call it callbuddy at somejabbersrv.com.
> This would be a bot that would accept commands - I've already made a small
> xmpp java bot which just logs into gtalk and send echo
> messages to users - like a parrot, nothing serious. Also I've set up
> openfire jabber server.
> What I would like him to do, is when user would tipe "call 0189432443" it
> would initiate a call between contact who tiped in this number (command)
> and this number.
>
> Is this possible, and what would be the best implementation (dingaling
> acting as client...)?
>
> Thanks.
>
> --
> kriko
>
>
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From brian at freeswitch.org  Wed Dec 17 09:21:32 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Dec 2008 11:21:32 -0600
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
In-Reply-To: <2bee4fc40812170915p1b5d91feu5fcfbee6713fad40@mail.gmail.com>
References: <2bee4fc40812170915p1b5d91feu5fcfbee6713fad40@mail.gmail.com>
Message-ID: <876638A9-FBB0-4079-A76F-814E6953D395@freeswitch.org>

FreeSWITCH already logs into your jabber server as a component if you  
cant communicate with other domains then your jabber server is not  
configured correctly.

/b

On Dec 17, 2008, at 11:15 AM, Kirk Bateman wrote:

> Kriko,
>
> I have been looking at the same sort of thing, but I'm planning to  
> implement an ejabberd bot component (so I can hopefully use the new  
> mod_erlang_event freeswitch interface).
>
> It seems to me that bits of the current dingaling / jingle interface  
> are having problems, like not liking sending messages to other  
> domains, its generally working if they are all .gmail.com users but  
> when you have some ongooglemail.com etc it starts breaking, and  
> doesn't use the whole JID in the from attribute for sending messages.
>
> When I get a chance I'll try and narrow down the problem.
>
> Cheers
>
> Kirk

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From msc at freeswitch.org  Wed Dec 17 09:33:19 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 17 Dec 2008 09:33:19 -0800
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <49491B0E.8080505@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
	<4947EBB2.7030401@ewetel.de> <49491272.9000103@ewetel.de>
	<191c3a030812170710u1f85fe97s7121cda8df02a50@mail.gmail.com>
	<49491B0E.8080505@ewetel.de>
Message-ID: <9C1B9E67-82A8-4FA0-BE62-01836FDC5594@freeswitch.org>

Helmut,

Can you turn on full debug and capture the output? It's a lot so put  
it in a pastebin.

-MC

Sent from my iPhone

On Dec 17, 2008, at 7:30 AM, Helmut Kuper <helmut.kuper at ewetel.de>  
wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> thx, but that doesn't work very good. Outgoing calls ring only once  
> and
> then this error rises in console:
>
> 2008-12-17 16:23:19 [DEBUG] Span:0 Q.931() Sending message to Layer4
> (size: 103)
> 2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I  
> got
> an event! Type:[02] Size:[103] CRV: 7 (0x7, CTX: Terminator)
> 2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan
> 810f590 (1:1) source isdn_data->channels_local_crv[0x7]
> 2008-12-17 16:23:19 [CRIT] ozmod_isdn.c:701 zap_isdn_931_34() Received
> CALL PROCEEDING message for channel 0
> 2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:702 zap_isdn_931_34()  
> Changing
> state on 1:1 from DIALING to PROGRESS
>
> regards
> Helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklJGw4ACgkQ4tZeNddg3dyeoACfeM6hYQF45T2gg18RQsOpZIjS
> SB0AoIQp+ixmWUGBKyMFXIZQ6AbQsWK0
> =I2L3
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From stephen at stephenjc.com  Wed Dec 17 10:00:09 2008
From: stephen at stephenjc.com (stephen at stephenjc)
Date: Wed, 17 Dec 2008 13:00:09 -0500
Subject: [Freeswitch-users] ignore dtmf
Message-ID: <1d9d102c0812171000i53a9e7d0u9557099f2b79cfe@mail.gmail.com>

I have a click to click system written in javascript, so i call out
both legs of the call then bridge them. I am looking for a way to
ignore dtmf tones on 1 leg of the call.


Thanks,
Stephen C
-All of my email addresses go to the same place
-Save Paper, think before you print.


From brian at freeswitch.org  Wed Dec 17 10:32:00 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Dec 2008 12:32:00 -0600
Subject: [Freeswitch-users] Cisco contest
In-Reply-To: <200812171541.mBHFfj4Y010761@box7.911domain.com>
References: <200812171541.mBHFfj4Y010761@box7.911domain.com>
Message-ID: <782B82A4-FF05-4F56-AB10-E8D6C6BD8A7C@freeswitch.org>

Seems pretty sleazy to me... what are they going to commercialize the  
results?

/b

On Dec 17, 2008, at 9:41 AM, freeswitch at davidnicol.otherinbox.com wrote:

> http://www.google.com/search?q=cisco+linux+contest
>
>
> Although cisco already does VOIP stuff so they might have trouble  
> awarding prizes to a technology which would compete with themselves,  
> but what are they expecting, putting Linux boards in Cisco backplanes?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mszlazak at aol.com  Wed Dec 17 11:29:36 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 17 Dec 2008 14:29:36 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
In-Reply-To: <674E25FB-0584-4357-B556-81BF79F84209@jerris.com>
References: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com><957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com><8CB2E358C58A406-9B8-1883@WEBMAIL-MA13.sysops.aol.com>
	<674E25FB-0584-4357-B556-81BF79F84209@jerris.com>
Message-ID: <8CB2E94EE5B5E49-E20-3F7@FWM-M37.sysops.aol.com>


 


 Hi Mike,

A brief talk with one of the Prophecy support people makes him think it maybe more a FreeSwitch issue but he wasn't that's been following my problem.

I've attached a pcap and FS log file from a past session. Also, there is a file with netstat's output.

The impression is that FS is not forwarding the audio to Prophecy correctly so Prophecy is timing out in different parts of it's dialogue before the hang up.

On the other hand, there is no problem when using "bypass media" but of course I can't use this if the media is being processed by FS before it's past on to.

Could please take a look.

Thanks. Mark. 


 

-----Original Message-----
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 17 Dec 2008 9:12 am
Subject: Re: [Freeswitch-users] Help with routing sound locally through FS









I think the best way to confirm all this is to load a full pcap in wireshark and have it pull the wav file of the individual audio streams to see what is going on.



Mike




On Dec 17, 2008, at 3:06 AM, mszlazak at aol.com wrote:



 Hi Mike,
 
 That does get the audio go between the softphone and the application (Voxeo's Prophecy ASR) "around" FreeSwitch but I would like the audio going "through" FreeSwitch. I plan to do something to it before passing it on.
 
 Support from Voxeo had this to say about the "bypass media" setting and if you could add some more insight that would be much appreciated. Since this is all on one Windows XP machine they can't get the info from the pcap file and are requesting I set up freeswitch on another machine which I will do. I thought you may have some more input.





=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 
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From itsc99 at cantv.net  Wed Dec 17 11:36:56 2008
From: itsc99 at cantv.net (Tjapko Smits)
Date: Wed, 17 Dec 2008 15:06:56 -0430
Subject: [Freeswitch-users] multi domain issue
Message-ID: <1229542616.24489.30.camel@tjaracas-main>

Hi, 

Fresh freeswitch user. Installed freeswitch with default installation a
week ago. Need more information on multi domain usage. Followed the wiki
pages with multi tenant examples. All working well but I do have a
problem when calling to an internal extension. 

Scenario: 

domain_A created by copying the default.xml to domain_A.xml following
steps from the wiki multi tenant informaion. Than created the domain_A
directory and copied the /default directory extensions there. 
Same steps for Domain_B. For the rest all is like default. 

Registered 3 phones to Domain_A and 3 phones to Domain_B 

Domain_A -> 1000 1005 and 1006 

Domain_B -> 1002 1005 and 1006

When I call from 1002 inside Domain_B to extension 1000 in Domain_A
phone rings and this is what I do not like to happen. 

Trace shows that when called to 1000 or 1005 or 1006 only the IP
addresses from those endpoints in Domain_A are addressed.

The first INVITE shows an OK invite with correct domain name but after
the 100 trying the re-invite makes an IP address out of most certainly
because it followed the rules in the dialplan.   

Can anybody point me in the direction how to make it possible that calls
from DOmain_A stay in Domain_A- and Domain_B stay in B etc. 

All the rest like inbound and outbound calls on dedicated gateways are
working fine. 

I assume that I need to configure acl for this but still not very clear.
Any tip will be most appreciated.  

-- 
Tjapko




From oseslija at gmail.com  Wed Dec 17 13:38:33 2008
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 17 Dec 2008 22:38:33 +0100
Subject: [Freeswitch-users] multi domain issue
In-Reply-To: <1229542616.24489.30.camel@tjaracas-main>
References: <1229542616.24489.30.camel@tjaracas-main>
Message-ID: <4468a6770812171338m153667ebnc567716e47dbc44f@mail.gmail.com>

Hi,

I have multi domain, multi tenant setup configured and working.

Did you add something like

    <param name="alias" value="sip:domain_A:5060"/>
    <param name="alias" value="sip:domain_B:5060"/>

to one of the profile configs for multi-domain so FreeSWITCH can look its
configs for those domains?
Also, check if you specified domain_A for "domain_name" param in the
domain_A.xml file.
Directory extensions context for other domains should not use the same
"user_context" param for they will hit default dialplan as well.

Please join the IRC channel (#freeswitch) for further questions you might
have.

Regards,
Ognjen
(sekil)
On Wed, Dec 17, 2008 at 8:36 PM, Tjapko Smits <itsc99 at cantv.net> wrote:

> Hi,
>
> Fresh freeswitch user. Installed freeswitch with default installation a
> week ago. Need more information on multi domain usage. Followed the wiki
> pages with multi tenant examples. All working well but I do have a
> problem when calling to an internal extension.
>
> Scenario:
>
> domain_A created by copying the default.xml to domain_A.xml following
> steps from the wiki multi tenant informaion. Than created the domain_A
> directory and copied the /default directory extensions there.
> Same steps for Domain_B. For the rest all is like default.
>
> Registered 3 phones to Domain_A and 3 phones to Domain_B
>
> Domain_A -> 1000 1005 and 1006
>
> Domain_B -> 1002 1005 and 1006
>
> When I call from 1002 inside Domain_B to extension 1000 in Domain_A
> phone rings and this is what I do not like to happen.
>
> Trace shows that when called to 1000 or 1005 or 1006 only the IP
> addresses from those endpoints in Domain_A are addressed.
>
> The first INVITE shows an OK invite with correct domain name but after
> the 100 trying the re-invite makes an IP address out of most certainly
> because it followed the rules in the dialplan.
>
> Can anybody point me in the direction how to make it possible that calls
> from DOmain_A stay in Domain_A- and Domain_B stay in B etc.
>
> All the rest like inbound and outbound calls on dedicated gateways are
> working fine.
>
> I assume that I need to configure acl for this but still not very clear.
> Any tip will be most appreciated.
>
> --
> Tjapko
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Dec 17 18:08:17 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 17 Dec 2008 18:08:17 -0800
Subject: [Freeswitch-users] ignore dtmf
In-Reply-To: <1d9d102c0812171000i53a9e7d0u9557099f2b79cfe@mail.gmail.com>
References: <1d9d102c0812171000i53a9e7d0u9557099f2b79cfe@mail.gmail.com>
Message-ID: <459F63E0-9E26-41DA-B5DD-FB95ACFC132C@freeswitch.org>

By ignore do you mean filter out? Or do you mean don't do anything but  
do audiblize the tones? Do you have some sort of application that does  
something with dtmfs?

-MC

Sent from my iPhone

On Dec 17, 2008, at 10:00 AM, "stephen at stephenjc" <stephen at stephenjc.com 
 > wrote:

> I have a click to click system written in javascript, so i call out
> both legs of the call then bridge them. I am looking for a way to
> ignore dtmf tones on 1 leg of the call.
>
>
> Thanks,
> Stephen C
> -All of my email addresses go to the same place
> -Save Paper, think before you print.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From stephen at stephenjc.com  Wed Dec 17 18:39:31 2008
From: stephen at stephenjc.com (stephen at stephenjc)
Date: Wed, 17 Dec 2008 21:39:31 -0500
Subject: [Freeswitch-users] ignore dtmf
In-Reply-To: <459F63E0-9E26-41DA-B5DD-FB95ACFC132C@freeswitch.org>
References: <1d9d102c0812171000i53a9e7d0u9557099f2b79cfe@mail.gmail.com>
	<459F63E0-9E26-41DA-B5DD-FB95ACFC132C@freeswitch.org>
Message-ID: <1d9d102c0812171839n3102ec33wf575fab7a9a73cfa@mail.gmail.com>

I have a click2click app that originates both legs of the call. After
the call is bridged if you press the # key it records the call. Only
the original person should be able to click # to enable records. That
is why i want to filter dtmf on 1 leg of the call.

Thanks,
Stephen C
-All of my email addresses go to the same place
-Save Paper, think before you print.




On Wed, Dec 17, 2008 at 9:08 PM, Michael S Collins <msc at freeswitch.org> wrote:
> By ignore do you mean filter out? Or do you mean don't do anything but
> do audiblize the tones? Do you have some sort of application that does
> something with dtmfs?
>
> -MC
>
> Sent from my iPhone
>
> On Dec 17, 2008, at 10:00 AM, "stephen at stephenjc" <stephen at stephenjc.com
>  > wrote:
>
>> I have a click to click system written in javascript, so i call out
>> both legs of the call then bridge them. I am looking for a way to
>> ignore dtmf tones on 1 leg of the call.
>>
>>
>> Thanks,
>> Stephen C
>> -All of my email addresses go to the same place
>> -Save Paper, think before you print.
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From scott.ellis at novatex.com.au  Wed Dec 17 18:45:26 2008
From: scott.ellis at novatex.com.au (Scott Ellis)
Date: Thu, 18 Dec 2008 13:45:26 +1100
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
In-Reply-To: <153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>
References: <49478B0F.3000802@novatex.com.au>
	<153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>
Message-ID: <4949B946.5050502@novatex.com.au>

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From markmorreny at gmail.com  Thu Dec 18 00:01:07 2008
From: markmorreny at gmail.com (mark morreny)
Date: Thu, 18 Dec 2008 16:01:07 +0800
Subject: [Freeswitch-users] Question about running Freeswitch in the
	background
Message-ID: <20ad6b920812180001t2ef2a2c7occ8130c57c69f3f5@mail.gmail.com>

Hi,

I have a small question about running Freeswich in the background using -nc
option.  Does freeswitch writes the log in any log file when running in
background mode?
If I have some print out statement in my custom mod, can I still see the
output of those logs somewhere?

Thanks,
Mark
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From jason at jasonjgw.net  Thu Dec 18 00:29:07 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 18 Dec 2008 19:29:07 +1100
Subject: [Freeswitch-users] Question about running Freeswitch in
	the	background
In-Reply-To: <20ad6b920812180001t2ef2a2c7occ8130c57c69f3f5@mail.gmail.com>
References: <20ad6b920812180001t2ef2a2c7occ8130c57c69f3f5@mail.gmail.com>
Message-ID: <20081218082907.GA11459@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 04:01:07PM +0800, mark morreny wrote:
> I have a small question about running Freeswich in the background using -nc
> option.  Does freeswitch writes the log in any log file when running in
> background mode?

Yes. Have a look at ~freeswitch/log/freeswitch.log




From kristjan.ugrin at gmail.com  Thu Dec 18 00:37:38 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 18 Dec 2008 09:37:38 +0100
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
In-Reply-To: <876638A9-FBB0-4079-A76F-814E6953D395@freeswitch.org>
References: <2bee4fc40812170915p1b5d91feu5fcfbee6713fad40@mail.gmail.com>
	<876638A9-FBB0-4079-A76F-814E6953D395@freeswitch.org>
Message-ID: <op.umcnkzr2atcbto@linux.site>

I'm not sure if we understood each other correctly.
I meant calling from jabber to other sip phones. I'm not sure you can just  
add a sip phone as
a buddy into e.g. gtalk or any other jabber service and call it.
So that why a bot (and only one but), which you would have as a buddy and  
you would feed him with numbers.
And he would estabilish a call between jabber user and typed in number.
Was I clearer?

I'll take a look how the component mode works, hopefully there is more  
documentation than only this:
http://wiki.freeswitch.org/wiki/Dingaling

Also if this functionality is not possible with fs as it is, I could maybe  
write a java program which interact with fs
interface and do that as intended?

Cheers

On Wed, 17 Dec 2008 18:21:32 +0100, Brian West <brian at freeswitch.org>  
wrote:

> FreeSWITCH already logs into your jabber server as a component if you
> cant communicate with other domains then your jabber server is not
> configured correctly.
>
> /b
>
> On Dec 17, 2008, at 11:15 AM, Kirk Bateman wrote:
>
>> Kriko,
>>
>> I have been looking at the same sort of thing, but I'm planning to
>> implement an ejabberd bot component (so I can hopefully use the new
>> mod_erlang_event freeswitch interface).
>>
>> It seems to me that bits of the current dingaling / jingle interface
>> are having problems, like not liking sending messages to other
>> domains, its generally working if they are all .gmail.com users but
>> when you have some ongooglemail.com etc it starts breaking, and
>> doesn't use the whole JID in the from attribute for sending messages.
>>
>> When I get a chance I'll try and narrow down the problem.
>>
>> Cheers
>>
>> Kirk
>



-- 
kriko


From jason at jasonjgw.net  Thu Dec 18 00:53:23 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 18 Dec 2008 19:53:23 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
	<5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
Message-ID: <20081218085323.GA11635@jdc.jasonjgw.net>

On Wed, Dec 17, 2008 at 12:10:18PM -0500, Michael Jerris wrote:
> 
> On Dec 17, 2008, at 2:40 AM, Jason White wrote:
> 
> > The code in bind6only_check in libs/sofia-sip/libsofia-sip-ua/tport/ 
> > tport.c looks
> > correct to me, but I can't find where the result is tested (it's in
> > mr_bindv6only). When bind6only_check() is called in  
> > tport_bind_server(), the
> > return value isn't tested, and I'm having difficulty finding where  
> > it is used

I realized after posting that if the IPv4 port is bound by another process,
then the attempt to bind to the IPv4 port in bind6only_check() should return
-1, and hence the result of bind6only_check() will be 0, even if the os allows
the IPv6 and IPv4 ports to be bound independently of each other.

This looks like a potential bug, but I haven't investigated properly to be
sure, and I'm extremely busy just now (as well as not being very experienced
at this).

> If this is in fact a bug, could you please report it to the sofia-sip  
> bugtracker.  Patches are very helpful there.

If anyone else has a chance to look at it before I do, please let me know.
Otherwise, I'll check it out when I have time to build a version of FreeSWITCH
with debug symbols and run it under gdb.



From brian at freeswitch.org  Thu Dec 18 01:01:12 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 03:01:12 -0600
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <20081218085323.GA11635@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
	<5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
	<20081218085323.GA11635@jdc.jasonjgw.net>
Message-ID: <0B5DCBAA-5FA5-488A-8189-39878A5FCDA0@freeswitch.org>

I bind mine independently without a problem on CentOS 5.2

/b

On Dec 18, 2008, at 2:53 AM, Jason White wrote:

> I realized after posting that if the IPv4 port is bound by another  
> process,
> then the attempt to bind to the IPv4 port in bind6only_check()  
> should return
> -1, and hence the result of bind6only_check() will be 0, even if the  
> os allows
> the IPv6 and IPv4 ports to be bound independently of each other.



From jason at jasonjgw.net  Thu Dec 18 01:11:49 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 18 Dec 2008 20:11:49 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <0B5DCBAA-5FA5-488A-8189-39878A5FCDA0@freeswitch.org>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
	<5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
	<20081218085323.GA11635@jdc.jasonjgw.net>
	<0B5DCBAA-5FA5-488A-8189-39878A5FCDA0@freeswitch.org>
Message-ID: <20081218091149.GA11826@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 03:01:12AM -0600, Brian West wrote:
> I bind mine independently without a problem on CentOS 5.2

Thanks; friends of mine have access to Fedora boxes, so we'll compare
behaviour and try to sort it out.



From fidibus83 at aol.com  Thu Dec 18 01:20:24 2008
From: fidibus83 at aol.com (fidibus83)
Date: Thu, 18 Dec 2008 10:20:24 +0100
Subject: [Freeswitch-users] SQL Error
Message-ID: <004801c960f1$dc680840$6445310a@Franzi>

Hello,

 

I?m a newbie in FS. I get an error from the freeswitch cli:

 

2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL ERR
[database disk image is malformed]

 

I don?t know what to do to remove this error! Can you help me?

 

Regards
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From hads at nice.net.nz  Thu Dec 18 01:37:33 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Thu, 18 Dec 2008 22:37:33 +1300
Subject: [Freeswitch-users] SQL Error
In-Reply-To: <004801c960f1$dc680840$6445310a@Franzi>
References: <004801c960f1$dc680840$6445310a@Franzi>
Message-ID: <200812182237.33739.hads@nice.net.nz>

On Thursday 18 December 2008 22:20:24 fidibus83 wrote:
> I?m a newbie in FS. I get an error from the freeswitch cli:
>
> 2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL
> ERR [database disk image is malformed]
>
> I don?t know what to do to remove this error! Can you help me?

If you remove the database files (which are in $PREFIX/db) then FreeSWITCH will 
recreate then at startup.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier


From carole.olivier at enst.fr  Thu Dec 18 01:37:42 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 18 Dec 2008 01:37:42 -0800 (PST)
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <49491847.3030403@lists.rupa.com>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
	<21054080.post@talk.nabble.com> <49491847.3030403@lists.rupa.com>
Message-ID: <21069519.post@talk.nabble.com>


Hello,

Thanks for your answers!
Concerning the creation of a new variable for the conference the problem is
that I do not create channels from the conference. I call separately a new
member on a new channel and add it on the conference only if he agrees to
enter it. So it was the same problem as for the uuid, I am not sure I can
access the good variable from anywhere in case many conferences are running.

So, I do the following if somebody is interested in:

<extension name="Local_Extension">
     .....................
      <condition field="destination_number"
expression="^${caller_id_number}$">
        <anti-action application="bind_meta_app" data="1 a o transfer::
transf_opp_leg_${db(select/conference/${caller_id_number})}_${destination_number}"/>
        <anti-action application="bind_meta_app" data="2 ab s
transfer::-both
transf_both_${uuid}_${caller_id_number}_${destination_number}"/>
       ..........
</extension>

<extension name="transf_both_legs">
     <condition field="destination_number"
expression="^transf_both_(.*)_(.*)_(.*)$">
         <action application="db" data="insert/conference/$2/conf_$1"/>
         <action application="db" data="insert/conference/$3/conf_$1"/>
         <action application="answer"/>
         <action application="conference" data="conf_$1 at conf_dynamic"/>
     </condition>
</extension>

<extension name="transf_opposite_leg">
     <condition field="destination_number"
expression="^transf_opp_leg_(.*)_(.*)$">
         <action application="db" data="insert/conference/$2/$1"/>
         <action application="answer"/>
         <action application="conference" data="$1 at conf_dynamic"/>
     </condition>
</extension>

This is not perfect and I am not sure it is "cleanly" programmed but it
works and it is flexible. All the members in the conference can invite new
members thanks to the conference name they all have in the database.

If you still have critics they are all welcome!
Thanks for your help,
Best regards,
Carole


Rupa Schomaker (lists)-2 wrote:
> 
> On 12/17/2008 8:24 AM, Carole O. wrote:
>> It would be unique you are right but I am not sure I can get its value if
>> A
>> puts the call on hold, calls C and wants to add it to the conference
>> whose
>> name dependent of the uuid of another session. 
>> I think if I use ${uuid} to add C I will have the uuid of the session
>> between A and C and not A and B no?
>> And I really have to configure this from the dialplan so statically.
>> 
>> Am I wrong somewhere??
>> 
>> Carole
> 
> Ah, yeah.  uuid would not be the same when initiating a new call that
> you then transfer to the conference call.  You need something that is
> intrinsic to the endpoint.
> 
> I did a quick info dump to an originated call.  Depending on your
> use-case (are these calls originating from registered handsets, trunked
> from a sip provider, etc) you might want to rely on the variable
> "sip_contact_uri" which is a combination of registered user name and ip
> (and port if port isn't 5060).  This should be unique per endpoint.
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21069519.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From kirk.bateman at gmail.com  Thu Dec 18 01:45:54 2008
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Thu, 18 Dec 2008 09:45:54 +0000
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
Message-ID: <2bee4fc40812180145mda36e7jf1b2905b9a5c63af@mail.gmail.com>

Brian,

That wasn't exactly what I meant :)

I have had Freeswitch connecting to GTalk directly as a client and that was
where I was getting the issues with sending anything to other domains.

I haven't actually tried the server profile with my own ejabberd server.
What I was planning to do was have a separate ejabberd bot which is the
control interface which then sends bits to the freeswitch connected "bot" /
user so I can sanitise anything before it gets to freeswitch, as I have
found the freeswitch jabber stuff to segfault every now and then (that said
I haven't updated in about a month or so). Just haven't had enough time to
actually look into what was causing the segfaults.

I did get the gtalk to sip bits working though, had a few codec issues but
gtalk to freeswitch worked fine, unfortunately my slightly more complicated
gtalk to freeswitch to asterisk to SPA941 has some codec bits to work out, I
think its to do with asterisk PCMU being different ? I know I had something
similar when trying to use the same setup with gizmo (which I fixed by
forcing G729 from SPA941 -> asterisk -> freeswitch.

Cheers

Kirk


> Date: Wed, 17 Dec 2008 11:21:32 -0600
> From: Brian West <brian at freeswitch.org>
> Subject: Re: [Freeswitch-users] Call sip phones from gtalk / jabber
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <876638A9-FBB0-4079-A76F-814E6953D395 at freeswitch.org>
> Content-Type: text/plain; charset="us-ascii"
>
> FreeSWITCH already logs into your jabber server as a component if you
> cant communicate with other domains then your jabber server is not
> configured correctly.
>
> /b
>
> On Dec 17, 2008, at 11:15 AM, Kirk Bateman wrote:
>
> > Kriko,
> >
> > I have been looking at the same sort of thing, but I'm planning to
> > implement an ejabberd bot component (so I can hopefully use the new
> > mod_erlang_event freeswitch interface).
> >
> > It seems to me that bits of the current dingaling / jingle interface
> > are having problems, like not liking sending messages to other
> > domains, its generally working if they are all .gmail.com users but
> > when you have some ongooglemail.com etc it starts breaking, and
> > doesn't use the whole JID in the from attribute for sending messages.
> >
> > When I get a chance I'll try and narrow down the problem.
> >
> > Cheers
> >
> > Kirk
>
>
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From fidibus83 at aol.com  Thu Dec 18 01:53:42 2008
From: fidibus83 at aol.com (fidibus83)
Date: Thu, 18 Dec 2008 10:53:42 +0100
Subject: [Freeswitch-users] SQL Error
In-Reply-To: <200812182237.33739.hads@nice.net.nz>
References: <004801c960f1$dc680840$6445310a@Franzi>
	<200812182237.33739.hads@nice.net.nz>
Message-ID: <006b01c960f6$83584030$6445310a@Franzi>

Thanks. It's ok again!

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Hadley
Rich
Gesendet: Donnerstag, 18. Dezember 2008 10:38
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] SQL Error

On Thursday 18 December 2008 22:20:24 fidibus83 wrote:
> I?m a newbie in FS. I get an error from the freeswitch cli:
>
> 2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL
> ERR [database disk image is malformed]
>
> I don?t know what to do to remove this error! Can you help me?

If you remove the database files (which are in $PREFIX/db) then FreeSWITCH
will 
recreate then at startup.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From kirk.bateman at gmail.com  Thu Dec 18 02:03:00 2008
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Thu, 18 Dec 2008 10:03:00 +0000
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
Message-ID: <2bee4fc40812180203u390f189ev970a7c714662185a@mail.gmail.com>

Kriko,

Have a look at this, I used it to get my gtalk to fs working.

http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/

Cheers

Kirk

Date: Thu, 18 Dec 2008 09:37:38 +0100
> From: kriko <kristjan.ugrin at gmail.com>
> Subject: Re: [Freeswitch-users] Call sip phones from gtalk / jabber
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <op.umcnkzr2atcbto at linux.site>
> Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8
>
> I'm not sure if we understood each other correctly.
> I meant calling from jabber to other sip phones. I'm not sure you can just
> add a sip phone as
> a buddy into e.g. gtalk or any other jabber service and call it.
> So that why a bot (and only one but), which you would have as a buddy and
> you would feed him with numbers.
> And he would estabilish a call between jabber user and typed in number.
> Was I clearer?
>
> I'll take a look how the component mode works, hopefully there is more
> documentation than only this:
> http://wiki.freeswitch.org/wiki/Dingaling
>
> Also if this functionality is not possible with fs as it is, I could maybe
> write a java program which interact with fs
> interface and do that as intended?
>
> Cheers
>
>
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From kristjan.ugrin at gmail.com  Thu Dec 18 02:39:13 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 18 Dec 2008 11:39:13 +0100
Subject: [Freeswitch-users] Java example
Message-ID: <op.umcs7ncwatcbto@linux.site>

I made a simple java example, following this guide
http://wiki.freeswitch.org/wiki/Java

so when someone calls it should print something in console.
I've also modified dialplan/public.xml, what I want is to intercept calls  
 from jabber and process them:
http://pastebin.com/m35de11d9

But there is nothing printed into console, path to jar is ok, also class  
name is ok.
Java module is being loaded on startup.

What could be wrong?

-- 
kriko


From damjan at ecntelecoms.com  Thu Dec 18 02:53:22 2008
From: damjan at ecntelecoms.com (damjan at ecntelecoms.com)
Date: Thu, 18 Dec 2008 12:53:22 +0200 (SAST)
Subject: [Freeswitch-users] Java example
In-Reply-To: <op.umcs7ncwatcbto@linux.site>
References: <op.umcs7ncwatcbto@linux.site>
Message-ID: <6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>

> I made a simple java example, following this guide
> http://wiki.freeswitch.org/wiki/Java
>
> so when someone calls it should print something in console.
> I've also modified dialplan/public.xml, what I want is to intercept calls
>  from jabber and process them:
> http://pastebin.com/m35de11d9
>
> But there is nothing printed into console, path to jar is ok, also class
> name is ok.
> Java module is being loaded on startup.
>
> What could be wrong?
>
> --
> kriko

You need to specify the path to the JAR file containing
si.marand.freeswitch.PhoneTest, which is definitely not freeswitch.jar.

Otherwise try attaching a remote debugger to the Java module and trace
through it as described on that wiki.

Damjan




From kristjan.ugrin at gmail.com  Thu Dec 18 04:09:48 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 18 Dec 2008 13:09:48 +0100
Subject: [Freeswitch-users] Java example
In-Reply-To: <6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>
References: <op.umcs7ncwatcbto@linux.site>
	<6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>
Message-ID: <op.umcxenzsatcbto@linux.site>

It is the right jar, I renamed it now to phoneTest.jar but still not  
working.
Do I have to specify whole path to the program inside class or just the  
class?
Remote debugging is working, but breakpoints never got triggered, so it is  
not being
executed at all.

I'm trying to process a call from gtalk to sip, maybe my public.xml is  
misconfigured (now I changed jar to phoneTest.jar):
http://pastebin.com/m35de11d9
(or public.xml it is not the right place at all?)


On Thu, 18 Dec 2008 11:53:22 +0100, <damjan at ecntelecoms.com> wrote:

> You need to specify the path to the JAR file containing
> si.marand.freeswitch.PhoneTest, which is definitely not freeswitch.jar.
>
> Otherwise try attaching a remote debugger to the Java module and trace
> through it as described on that wiki.
>
> Damjan
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
kriko


From mike at jerris.com  Thu Dec 18 05:10:19 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Dec 2008 08:10:19 -0500
Subject: [Freeswitch-users] Question about running Freeswitch in the
	background
In-Reply-To: <20081218082907.GA11459@jdc.jasonjgw.net>
References: <20ad6b920812180001t2ef2a2c7occ8130c57c69f3f5@mail.gmail.com>
	<20081218082907.GA11459@jdc.jasonjgw.net>
Message-ID: <74FAFF44-C6F1-4AF8-A0C9-12DCDD16992C@jerris.com>

A note, logging is handled by mod_logfile, it has nothing to do if you  
run in the background or not.

Mike

On Dec 18, 2008, at 3:29 AM, Jason White wrote:

> On Thu, Dec 18, 2008 at 04:01:07PM +0800, mark morreny wrote:
>> I have a small question about running Freeswich in the background  
>> using -nc
>> option.  Does freeswitch writes the log in any log file when  
>> running in
>> background mode?
>
> Yes. Have a look at ~freeswitch/log/freeswitch.log
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Thu Dec 18 05:12:03 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Dec 2008 08:12:03 -0500
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
In-Reply-To: <2bee4fc40812180145mda36e7jf1b2905b9a5c63af@mail.gmail.com>
References: <2bee4fc40812180145mda36e7jf1b2905b9a5c63af@mail.gmail.com>
Message-ID: <F3079C63-BB3B-4487-B458-2B941B49E91C@jerris.com>


On Dec 18, 2008, at 4:45 AM, Kirk Bateman wrote:

>  have had Freeswitch connecting to GTalk directly as a client and  
> that was where I was getting the issues with sending anything to  
> other domains.

I have seen this before specifically with gmail being unable to  
federate presense... their stuff can be really flakey, do you see the  
same thing if you client login to a jabber.org address?



From yudha2008 at gmail.com  Thu Dec 18 05:13:08 2008
From: yudha2008 at gmail.com (Baskar)
Date: Thu, 18 Dec 2008 18:43:08 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
Message-ID: <d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>

*Hi,

I am using JavaScript file to detect busy tone signals but I cant able to
detect the busy tone signals
*
*My JavaScript*
*
session1 = new Session();
session1.originate(session1, "{ignore_early_media=true}sofia/default/
39841799874 at 172.20.191.228");
session1.execute("tone_detect", "busy 400 r");
session1.execute("bridge", "sofia/default/39841799874 at 172.20.191.228");
session1.execute("transfer", "39841799874");*


*I get output:*

*freeswitch at localhost.localdomain> jsrun tone.js*
API CALL [jsrun(tone.js)] output:
OK

freeswitch at localhost.localdomain> 2008-12-18 18:42:30 [NOTICE]
switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/
39841799874 at 172.20.191.228 [0a9723ca-d170-4cee-a8bf-58a8ad018a44]
2008-12-18 18:42:35 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Ring-Ready sofia/internal/39841799874 at 172.20.191.228!
2008-12-18 18:42:35 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Pre-Answer sofia/internal/39841799874 at 172.20.191.228!
2008-12-18 18:42:39 [NOTICE] sofia.c:2963 sofia_handle_sip_i_state() Channel
[sofia/internal/39841799874 at 172.20.191.228] has been answered
2008-12-18 18:42:39 [NOTICE] mod_dptools.c:1217
tone_detect_session_function()* Enabling tone detection 'busy' '400'*
2008-12-18 18:42:39 [NOTICE] switch_channel.c:565 switch_channel_set_name()
New Channel sofia/internal/39841799874 at 172.20.191.228[5152416f-7e5c-4a60-9601-6a4af625d8aa]
2008-12-18 18:42:39 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Ring-Ready sofia/internal/39841799874 at 172.20.191.228!
2008-12-18 18:42:39 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Pre-Answer sofia/internal/39841799874 at 172.20.191.228!
*
when i run my js call is connected and after the caller answer only enabling
the tone detection.

I am not sure i am correct. correct me how to detect the busy signal. I have
written a small JavaScript. Correct me where i am wrong (In the program or
in the way it detect the call).

Thanks in advance.
*
*--
Warm Regards,
N.Baskar
*
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From anthony.minessale at gmail.com  Thu Dec 18 06:12:09 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Dec 2008 08:12:09 -0600
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
In-Reply-To: <4949B946.5050502@novatex.com.au>
References: <49478B0F.3000802@novatex.com.au>
	<153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>
	<4949B946.5050502@novatex.com.au>
Message-ID: <191c3a030812180612n94f9b72vc9ad0ba2d90d6a9e@mail.gmail.com>

can you press f8 to set the FS console to DEBUG and take the same capture.


On Wed, Dec 17, 2008 at 8:45 PM, Scott Ellis <scott.ellis at novatex.com.au>wrote:

>  After further checking, it does not seem like the authentication after the
> challenge is being sent...
>
> Are there any other settings I should be aware of other than placing the
> file in external and setting register to true?
>
> Scott
>
> 2008-12-18 13:32:28 [NOTICE] sofia_reg.c:265 sofia_reg_check_gateway()
> Registering sip.pennytel.com
> nua: nua_handle_bind: entering
> nua: nua_register: entering
> nua(0x89b08e0): sent signal r_register
> nua(0x89b08e0): recv signal r_register
> nua: nua_stack_set_params: entering
> soa_clone(static::0x8977798, 0x89792f8, 0x89b08e0) called
> soa_set_params(static::0x89c52c0, ...) called
> soa_set_params(static::0x89c52c0, ...) called
> nta_leg_tcreate(0x89c4948)
> nua(0x89b08e0): adding register usage
> nta: selecting scheme sip
> nta: for "sip.pennytel.com" query "_sip._udp.sip.pennytel.com" SRV
> nta: for "sip.pennytel.com" query "sip.pennytel.com" A (cached)
> nta: sip.pennytel.com. IN A 202.85.243.87
> tport_tsend(0x8976740) tpn = udp/202.85.243.87:5060
> tport_resolve addrinfo = 202.85.243.87:5060
> tport_by_addrinfo(0x8976740): not found by name udp/202.85.243.87:5060
> tport_vsend(0x8976740): 646 bytes of 646 to udp/202.85.243.87:5060
> tport_vsend returned 646
> send 646 bytes to udp/[202.85.243.87]:5060 at 02:32:30.322198:
>    ------------------------------------------------------------------------
>    REGISTER sip:sip.pennytel.com;transport=udp SIP/2.0
>    Via: SIP/2.0/UDP 203.113.255.140:5080;rport;branch=z9hG4bKt232eUFUNXr2e
>    Max-Forwards: 70
>    From: <sip:8xxxxx75 at sip.pennytel.com;transport=udp>;tag=t0Umc83St29ND
>    To: <sip:8xxxxxx75 at sip.pennytel.com;transport=udp>
>    Call-ID: d25d6f36-ccab-11dd-900f-67e92a02be7d
>    CSeq: 108665407 REGISTER
>    Contact: <sip:8xxxxxxx75 at 203.113.255.140:5080;transport=udp>
>    Expires: 600
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10760
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
>    Supported: timer, precondition, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> nta: sent REGISTER (108665407) to udp/202.85.243.87:5060
> tport_pend(0x8976740): pending 0x89f2d50 for udp/192.168.0.5:5080 (already
> 0)
> nta: timer set to 32000 ms
> nta: timer shortened to 500 ms
> tport_wakeup_pri(0x8976740): events IN
> tport_recv_event(0x8976740)
> tport_recv_iovec(0x8976740) msg 0x89eeeb8 from (udp/192.168.0.5:5080) has
> 518 bytes, veclen = 1
> recv 518 bytes from udp/[202.85.243.87]:5060 at 02:32:30.370072:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 203.113.255.140:5080
> ;rport=5080;branch=z9hG4bKt232eUFUNXr2e
>    From: <sip:8xxxxx75 at sip.pennytel.com;transport=udp>;tag=t0Umc83St29ND
>    To: <sip:8xxxxx75 at sip.pennytel.com;transport=udp>
> ;tag=abda4710fbd488d9ce6d01bba5c3e23b-cec7
>    Call-ID: d25d6f36-ccab-11dd-900f-67e92a02be7d
>    CSeq: 108665407 REGISTER
>    WWW-Authenticate: Digest realm="sip.pennytel.com",
> nonce="4949b76bf622961d78acb213b5556104938ecd6e"
>    Server: Sip EXpress router (0.9.6 (i386/freebsd))
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> tport_deliver(0x8976740): msg 0x89eeeb8 (518 bytes) from udp/
> 202.85.243.87:5080/sip next=(nil)
> nta: received 401 Unauthorized for REGISTER (108665407)
> nta: 401 Unauthorized is going to a transaction
> nta_outgoing: RTT is 49.89 ms
> tport_release(0x8976740): 0x89f2d50 by 0x89a4640 with 0x89eeeb8
> nta: timer set next to 4531 ms
> nta: timer K fired, terminate REGISTER (108665407)
> outgoing_reclaim_all((nil), (nil), 0xb2c6d1e8)
> nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/2 free
> nta: timer not set
> 2008-12-18 13:32:59 [WARNING] sofia_reg.c:307 sofia_reg_check_gateway()
> sip.pennytel.com Failed Registration, setting retry to 60 seconds.
>
>
> Michael Jerris wrote:
>
> We send authentication after we get a challenge because on startup we
> need the nonce from them to build the hash in the Auth header properly.
>
> Mike
>
> On Dec 16, 2008, at 6:03 AM, Scott Ellis wrote:
>
>
>
>  I have a standard install, and I am trying to get a Pennytel gateway
> to
> register.
>
> After looking at Wireshark traces of x-lite registering and FreeSwitch
> registering, FreeSwitch is not sending any authentication information
> with the registration request. I am obviously missing something here!
>
> I understand for incoming calls you don't want authentication, but for
> outgoing it is obviously required.
>
> Is there a flag somewhere that I am supposed to set? The file was
> taken
> from the wiki page, and looks like it was previously tested when using
> the obsolete outbound directory structure.
>
> The following file is in the conf/sip_profiles/external directory.
>
> <include>
>  <gateway name="PennyTel">
>    <param name="username" value="8xxxxxxx"/>
>    <param name="password" value="xxxxxxx"/>
>    <param name="realm" value="sip.pennytel.com"/>
>    <param name="proxy" value="sip.pennytel.com"/>
>    <param name="register" value="true"/>
>    <param name="expire-seconds" value="60"/>
>  </gateway>
> </include>
>
> Thanks.
>
> Scott
>
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>  _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Dec 18 06:19:22 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Dec 2008 08:19:22 -0600
Subject: [Freeswitch-users] Java example
In-Reply-To: <op.umcxenzsatcbto@linux.site>
References: <op.umcs7ncwatcbto@linux.site>
	<6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>
	<op.umcxenzsatcbto@linux.site>
Message-ID: <191c3a030812180619k394a9e33g7cba5808e7d95558@mail.gmail.com>

did you turn up your console log level high enough to see it? The default
level is "INFO"


On Thu, Dec 18, 2008 at 6:09 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> It is the right jar, I renamed it now to phoneTest.jar but still not
> working.
> Do I have to specify whole path to the program inside class or just the
> class?
> Remote debugging is working, but breakpoints never got triggered, so it is
> not being
> executed at all.
>
> I'm trying to process a call from gtalk to sip, maybe my public.xml is
> misconfigured (now I changed jar to phoneTest.jar):
> http://pastebin.com/m35de11d9
> (or public.xml it is not the right place at all?)
>
>
> On Thu, 18 Dec 2008 11:53:22 +0100, <damjan at ecntelecoms.com> wrote:
>
> > You need to specify the path to the JAR file containing
> > si.marand.freeswitch.PhoneTest, which is definitely not freeswitch.jar.
> >
> > Otherwise try attaching a remote debugger to the Java module and trace
> > through it as described on that wiki.
> >
> > Damjan
> >
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> kriko
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From peder at networkoblivion.com  Thu Dec 18 06:38:34 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Thu, 18 Dec 2008 08:38:34 -0600
Subject: [Freeswitch-users] Core Dump
Message-ID: <494A606A.2000601@networkoblivion.com>

What is the process for capturing and submitting a core dump?

I am messing around with the Cisco 79x1 phones and tcp and multiple reg. 
  I have a 7961 using tcp and a 7960 using udp both reg'd with the same 
number and both showing up as registered.  If I call out from the phone 
using tcp, it works.  If I call out from the phone using udp, I get a 
core dump.  If I call in, it calls both phones and I am assuming the 
call to the phone using udp causes a core dump as well.  These are the 
only two phones on the system and I am running version 10851 from 
yesterday.  If I only have the udp phone registered and the tcp phone is 
off, it works fine.  It is only when I have a mix of a udp reg and a tcp 
reg on the same number that I appear to get a core dump.


Peder


From msc at freeswitch.org  Thu Dec 18 06:43:06 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 18 Dec 2008 06:43:06 -0800
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
Message-ID: <980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>

You've got ignore_early_media set to true but busy signals might be  
sent during early media. Why are you ignoring early media?

Also, you might need to check your tone_detect syntax. You're set to  
detect 400Hz but you haven't told the system what to do if it does  
detect that tone. Please look at the wiki examples for tone_detect.  
You will see what kinds of things you can do with it, but usually you  
just set a channel variable.

-MC

Sent from my iPhone

On Dec 18, 2008, at 5:13 AM, Baskar <yudha2008 at gmail.com> wrote:

> Hi,
>
> I am using JavaScript file to detect busy tone signals but I cant  
> able to detect the busy tone signals
>
> My JavaScript
>
> session1 = new Session();
> session1.originate(session1, "{ignore_early_media=true}sofia/default/39841799874 at 172.20.191.228 
> ");
> session1.execute("tone_detect", "busy 400 r");
> session1.execute("bridge", "sofia/default/ 
> 39841799874 at 172.20.191.228");
> session1.execute("transfer", "39841799874");
>
>
> I get output:
>
> freeswitch at localhost.localdomain> jsrun tone.js
> API CALL [jsrun(tone.js)] output:
> OK
>
> freeswitch at localhost.localdomain> 2008-12-18 18:42:30 [NOTICE]  
> switch_channel.c:565 switch_channel_set_name() New Channel sofia/ 
> internal/39841799874 at 172.20.191.228 [0a9723ca-d170-4cee- 
> a8bf-58a8ad018a44]
> 2008-12-18 18:42:35 [NOTICE] sofia_glue.c:2097  
> sofia_glue_tech_media() Ring-Ready sofia/internal/39841799874 at 172.20.191.228 
> !
> 2008-12-18 18:42:35 [NOTICE] sofia_glue.c:2097  
> sofia_glue_tech_media() Pre-Answer sofia/internal/39841799874 at 172.20.191.228 
> !
> 2008-12-18 18:42:39 [NOTICE] sofia.c:2963 sofia_handle_sip_i_state()  
> Channel [sofia/internal/39841799874 at 172.20.191.228] has been answered
> 2008-12-18 18:42:39 [NOTICE] mod_dptools.c:1217  
> tone_detect_session_function() Enabling tone detection 'busy' '400'
> 2008-12-18 18:42:39 [NOTICE] switch_channel.c:565  
> switch_channel_set_name() New Channel sofia/internal/39841799874 at 172.20.191.228 
>  [5152416f-7e5c-4a60-9601-6a4af625d8aa]
> 2008-12-18 18:42:39 [NOTICE] sofia_glue.c:2097  
> sofia_glue_tech_media() Ring-Ready sofia/internal/39841799874 at 172.20.191.228 
> !
> 2008-12-18 18:42:39 [NOTICE] sofia_glue.c:2097  
> sofia_glue_tech_media() Pre-Answer sofia/internal/39841799874 at 172.20.191.228 
> !
>
> when i run my js call is connected and after the caller answer only  
> enabling the tone detection.
>
> I am not sure i am correct. correct me how to detect the busy  
> signal. I have written a small JavaScript. Correct me where i am  
> wrong (In the program or in the way it detect the call).
>
> Thanks in advance.
>
> -- 
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From fidibus83 at aol.com  Thu Dec 18 06:43:19 2008
From: fidibus83 at aol.com (fidibus83)
Date: Thu, 18 Dec 2008 15:43:19 +0100
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21069519.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com>
	<49490301.8040701@lists.rupa.com><21054080.post@talk.nabble.com>
	<49491847.3030403@lists.rupa.com> <21069519.post@talk.nabble.com>
Message-ID: <012501c9611e$f8e43bb0$6445310a@Franzi>

Hello,

Carole, your conference-programm is what I looked for. It's great. I try it.
But I get an Error, when I press *1 and I don't know what it mean... Have
you an idea?

[ERR] mod_conference.c:4849 conference_new() invalid Record! No name.
[CRIT] mod_conference.c:4314 conference_function() Memory Error!

Thanks! 

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Carole
O.
Gesendet: Donnerstag, 18. Dezember 2008 10:38
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] dynamic conference


Hello,

Thanks for your answers!
Concerning the creation of a new variable for the conference the problem is
that I do not create channels from the conference. I call separately a new
member on a new channel and add it on the conference only if he agrees to
enter it. So it was the same problem as for the uuid, I am not sure I can
access the good variable from anywhere in case many conferences are running.

So, I do the following if somebody is interested in:

<extension name="Local_Extension">
     .....................
      <condition field="destination_number"
expression="^${caller_id_number}$">
        <anti-action application="bind_meta_app" data="1 a o transfer::
transf_opp_leg_${db(select/conference/${caller_id_number})}_${destination_nu
mber}"/>
        <anti-action application="bind_meta_app" data="2 ab s
transfer::-both
transf_both_${uuid}_${caller_id_number}_${destination_number}"/>
       ..........
</extension>

<extension name="transf_both_legs">
     <condition field="destination_number"
expression="^transf_both_(.*)_(.*)_(.*)$">
         <action application="db" data="insert/conference/$2/conf_$1"/>
         <action application="db" data="insert/conference/$3/conf_$1"/>
         <action application="answer"/>
         <action application="conference" data="conf_$1 at conf_dynamic"/>
     </condition>
</extension>

<extension name="transf_opposite_leg">
     <condition field="destination_number"
expression="^transf_opp_leg_(.*)_(.*)$">
         <action application="db" data="insert/conference/$2/$1"/>
         <action application="answer"/>
         <action application="conference" data="$1 at conf_dynamic"/>
     </condition>
</extension>

This is not perfect and I am not sure it is "cleanly" programmed but it
works and it is flexible. All the members in the conference can invite new
members thanks to the conference name they all have in the database.

If you still have critics they are all welcome!
Thanks for your help,
Best regards,
Carole


Rupa Schomaker (lists)-2 wrote:
> 
> On 12/17/2008 8:24 AM, Carole O. wrote:
>> It would be unique you are right but I am not sure I can get its value if
>> A
>> puts the call on hold, calls C and wants to add it to the conference
>> whose
>> name dependent of the uuid of another session. 
>> I think if I use ${uuid} to add C I will have the uuid of the session
>> between A and C and not A and B no?
>> And I really have to configure this from the dialplan so statically.
>> 
>> Am I wrong somewhere??
>> 
>> Carole
> 
> Ah, yeah.  uuid would not be the same when initiating a new call that
> you then transfer to the conference call.  You need something that is
> intrinsic to the endpoint.
> 
> I did a quick info dump to an originated call.  Depending on your
> use-case (are these calls originating from registered handsets, trunked
> from a sip provider, etc) you might want to rely on the variable
> "sip_contact_uri" which is a combination of registered user name and ip
> (and port if port isn't 5060).  This should be unique per endpoint.
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context:
http://www.nabble.com/dynamic-conference-tp21053181p21069519.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From kristjan.ugrin at gmail.com  Thu Dec 18 06:56:11 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 18 Dec 2008 15:56:11 +0100
Subject: [Freeswitch-users] Gtalk to sip problems when reconfiguring from
	scratch
Message-ID: <op.umc43xs3atcbto@linux.site>

I recently purged all freeswitch config and restarted configuring from  
scratch. Using defaults, I modified public.xml dialplan config (added line  
16 - 28):
http://pastebin.com/m5ece6e6f

and added a new config under jingle_profiles:
http://pastebin.com/d6e983b99

I register with phonelite or twinkle as user 1000 (it says successfull  
reg.), but when I make a call from gtalk I hear that user is not  
available, none of
client rings.

When registering sofia prints:
2008-12-18 15:53:32 [INFO] sofia_presence.c:475  
actual_sofia_presence_event_handler() internal START_PRESENCE_PROBE_SQL
2008-12-18 15:53:32 [NOTICE] sofia_presence.c:793  
sofia_presence_resub_callback() internal PRESENCE_PROBE 1000 at 10.99.8.221
2008-12-18 15:53:32 [INFO] sofia_presence.c:484  
actual_sofia_presence_event_handler() internal END_PRESENCE_PROBE_SQL

2008-12-18 15:53:32 [INFO] sofia_presence.c:547  
actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal)
2008-12-18 15:53:32 [INFO] sofia_presence.c:563  
actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal)
2008-12-18 15:53:32 [WARNING] sofia_presence.c:517  
actual_sofia_presence_event_handler() external is passive, skipping
2008-12-18 15:53:32 [WARNING] sofia_presence.c:510  
actual_sofia_presence_event_handler() 10.99.8.221 is an alias, skipping
2008-12-18 15:53:32 [WARNING] sofia_presence.c:510  
actual_sofia_presence_event_handler() default is an alias, skipping
2008-12-18 15:53:32 [WARNING] sofia_presence.c:510  
actual_sofia_presence_event_handler() nat is an alias, skipping
2008-12-18 15:53:32 [WARNING] sofia_presence.c:510  
actual_sofia_presence_event_handler() outbound is an alias, skipping

and when fetching registrations:
2008-12-18 15:53:50 [ERR] sofia_reg.c:1120  
sofia_reg_handle_sip_i_register() NO CONTACT!

Does this means that registration failed? Why it doesn't call anymore the  
targeted user (1000)?

Here is also an extract from console log while calling, it caught my  
attention:
2008-12-18 15:54:41 [WARNING] mod_dptools.c:2047 user_outgoing_channel()  
Can't find user [1000@]
2008-12-18 15:54:41 [ERR] switch_ivr_originate.c:1110  
switch_ivr_originate() Cannot create outgoing channel of type [user]  
cause: [SUBSCRIBER_ABSENT]
2008-12-18 15:54:41 [DEBUG] switch_ivr_originate.c:1689  
switch_ivr_originate() Originate Resulted in Error Cause: 20  
[SUBSCRIBER_ABSENT]

-- 
kriko


From msc at freeswitch.org  Thu Dec 18 07:02:18 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 18 Dec 2008 07:02:18 -0800
Subject: [Freeswitch-users] Core Dump
Message-ID: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>

Check out this page:
wiki.freeswitch.org/wiki/Debugging_Freeswitch

-MC

Sent from my iPhone

On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com 
 > wrote:

> What is the process for capturing and submitting a core dump?
>
> I am messing around with the Cisco 79x1 phones and tcp and multiple  
> reg.
> I have a 7961 using tcp and a 7960 using udp both reg'd with the same
> number and both showing up as registered.  If I call out from the  
> phone
> using tcp, it works.  If I call out from the phone using udp, I get a
> core dump.  If I call in, it calls both phones and I am assuming the
> call to the phone using udp causes a core dump as well.  These are the
> only two phones on the system and I am running version 10851 from
> yesterday.  If I only have the udp phone registered and the tcp  
> phone is
> off, it works fine.  It is only when I have a mix of a udp reg and a  
> tcp
> reg on the same number that I appear to get a core dump.
>
>
> Peder
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From peder at networkoblivion.com  Thu Dec 18 07:36:00 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Thu, 18 Dec 2008 09:36:00 -0600
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
Message-ID: <494A6DE0.4030701@networkoblivion.com>

If anybody wants to look at the core dump in gdb, here it is (the actual 
core is 256Meg):

http://pastebin.freeswitch.org/6476

I know zip about debugging and gdb, but from looking through it, I see a 
segmentation fault and it appears to be thread 15094.  The last three 
items in the bt full for that thread are:

destroy_status = <value optimized out>
fd = (switch_file_t *) 0x80529b0
pool = (switch_memory_pool_t *) 0x80528f0

So I would guess it is trying to access an invalid memory location, but 
why, I have no idea....

Any ideas?


Peder


Michael S Collins wrote:
> Check out this page:
> wiki.freeswitch.org/wiki/Debugging_Freeswitch
> 
> -MC
> 
> Sent from my iPhone
> 
> On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com 
>  > wrote:
> 
>> What is the process for capturing and submitting a core dump?
>>
>> I am messing around with the Cisco 79x1 phones and tcp and multiple  
>> reg.
>> I have a 7961 using tcp and a 7960 using udp both reg'd with the same
>> number and both showing up as registered.  If I call out from the  
>> phone
>> using tcp, it works.  If I call out from the phone using udp, I get a
>> core dump.  If I call in, it calls both phones and I am assuming the
>> call to the phone using udp causes a core dump as well.  These are the
>> only two phones on the system and I am running version 10851 from
>> yesterday.  If I only have the udp phone registered and the tcp  
>> phone is
>> off, it works fine.  It is only when I have a mix of a udp reg and a  
>> tcp
>> reg on the same number that I appear to get a core dump.
>>
>>
>> Peder
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From msc at freeswitch.org  Thu Dec 18 08:09:50 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 18 Dec 2008 08:09:50 -0800
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <494A6DE0.4030701@networkoblivion.com>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<494A6DE0.4030701@networkoblivion.com>
Message-ID: <CAEDC248-D670-4603-A4A8-5F06074304E9@freeswitch.org>

Is this a single occurrence or can you make it happen consistently?
-MC

Sent from my iPhone

On Dec 18, 2008, at 7:36 AM, "peder at networkoblivion.com" <peder at networkoblivion.com 
 > wrote:

> If anybody wants to look at the core dump in gdb, here it is (the  
> actual
> core is 256Meg):
>
> http://pastebin.freeswitch.org/6476
>
> I know zip about debugging and gdb, but from looking through it, I  
> see a
> segmentation fault and it appears to be thread 15094.  The last three
> items in the bt full for that thread are:
>
> destroy_status = <value optimized out>
> fd = (switch_file_t *) 0x80529b0
> pool = (switch_memory_pool_t *) 0x80528f0
>
> So I would guess it is trying to access an invalid memory location,  
> but
> why, I have no idea....
>
> Any ideas?
>
>
> Peder
>
>
> Michael S Collins wrote:
>> Check out this page:
>> wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com
>>> wrote:
>>
>>> What is the process for capturing and submitting a core dump?
>>>
>>> I am messing around with the Cisco 79x1 phones and tcp and multiple
>>> reg.
>>> I have a 7961 using tcp and a 7960 using udp both reg'd with the  
>>> same
>>> number and both showing up as registered.  If I call out from the
>>> phone
>>> using tcp, it works.  If I call out from the phone using udp, I  
>>> get a
>>> core dump.  If I call in, it calls both phones and I am assuming the
>>> call to the phone using udp causes a core dump as well.  These are  
>>> the
>>> only two phones on the system and I am running version 10851 from
>>> yesterday.  If I only have the udp phone registered and the tcp
>>> phone is
>>> off, it works fine.  It is only when I have a mix of a udp reg and a
>>> tcp
>>> reg on the same number that I appear to get a core dump.
>>>
>>>
>>> Peder
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Thu Dec 18 08:21:11 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 10:21:11 -0600
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <494A6DE0.4030701@networkoblivion.com>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<494A6DE0.4030701@networkoblivion.com>
Message-ID: <C9C3FF52-4FA8-4241-8DDB-7672F194162D@freeswitch.org>

Peder,
	Can you join us on IRC.
/b

On Dec 18, 2008, at 9:36 AM, peder at networkoblivion.com wrote:

> If anybody wants to look at the core dump in gdb, here it is (the  
> actual
> core is 256Meg):
>
> http://pastebin.freeswitch.org/6476
>
> I know zip about debugging and gdb, but from looking through it, I  
> see a
> segmentation fault and it appears to be thread 15094.  The last three
> items in the bt full for that thread are:
>
> destroy_status = <value optimized out>
> fd = (switch_file_t *) 0x80529b0
> pool = (switch_memory_pool_t *) 0x80528f0
>
> So I would guess it is trying to access an invalid memory location,  
> but
> why, I have no idea....
>
> Any ideas?
>
>
> Peder
>
>
> Michael S Collins wrote:
>> Check out this page:
>> wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com
>>> wrote:
>>
>>> What is the process for capturing and submitting a core dump?
>>>
>>> I am messing around with the Cisco 79x1 phones and tcp and multiple
>>> reg.
>>> I have a 7961 using tcp and a 7960 using udp both reg'd with the  
>>> same
>>> number and both showing up as registered.  If I call out from the
>>> phone
>>> using tcp, it works.  If I call out from the phone using udp, I  
>>> get a
>>> core dump.  If I call in, it calls both phones and I am assuming the
>>> call to the phone using udp causes a core dump as well.  These are  
>>> the
>>> only two phones on the system and I am running version 10851 from
>>> yesterday.  If I only have the udp phone registered and the tcp
>>> phone is
>>> off, it works fine.  It is only when I have a mix of a udp reg and a
>>> tcp
>>> reg on the same number that I appear to get a core dump.
>>>
>>>
>>> Peder
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From odermann at googlemail.com  Thu Dec 18 08:36:32 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 17:36:32 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
Message-ID: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>

i would like to know, what the best way is, to redirect an incoming
call from one fs (fs1) to another fs (fs2).

we use 3 freeswitch servers and the carrier passes calls to the three
fs servers randomly. if on fs server is not offline, the carrier sends
the call to the next fs.
this is generally good, but for conferencing it not so good.

i am using socket outbound and need to do this for conferencing. let's
say, we have a conference going on on fs1. another person wants to
enter this conference, but the call is passed to fs2. on fs2 we see,
that the caller wants to enter the conference going on on fs1.

now we have to redirect the call from fs2 to fs1. is this done with
"redirect" and some according settings/params or are there other ways
to do this? we would like to do this without our carrier doing
something, to be a little more independant.

thanks
dennis


From brian at freeswitch.org  Thu Dec 18 08:43:25 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 10:43:25 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
Message-ID: <283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>

the deflect app.

/b

On Dec 18, 2008, at 10:36 AM, Dennis wrote:

> i would like to know, what the best way is, to redirect an incoming
> call from one fs (fs1) to another fs (fs2).



From odermann at googlemail.com  Thu Dec 18 08:52:23 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 17:52:23 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
Message-ID: <5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>

i had a look at the deflect app, but as far as i understand it, the
carrier has to support/understand it ans react on the signals.

is that right or does this have nothing to do with our carrier? or
does this work between fs servers in the same local network?


another similar question is: how to reject calls, so that the carrier
tries to route the call to another fs? if we want to make changes to
fs and test them, we would like to block new incoming calls, till
there are no running calls, to shut down the fs.




2008/12/18 Brian West <brian at freeswitch.org>:
> the deflect app.
>
> /b
>
> On Dec 18, 2008, at 10:36 AM, Dennis wrote:
>
>> i would like to know, what the best way is, to redirect an incoming
>> call from one fs (fs1) to another fs (fs2).
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Dec 18 08:58:24 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 10:58:24 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
Message-ID: <704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>

What switch is your provider using?

/b

On Dec 18, 2008, at 10:52 AM, Dennis wrote:

> i had a look at the deflect app, but as far as i understand it, the
> carrier has to support/understand it ans react on the signals.
>
> is that right or does this have nothing to do with our carrier? or
> does this work between fs servers in the same local network?
>
>
> another similar question is: how to reject calls, so that the carrier
> tries to route the call to another fs? if we want to make changes to
> fs and test them, we would like to block new incoming calls, till
> there are no running calls, to shut down the fs.



From jonas.gauffin at gmail.com  Thu Dec 18 09:05:06 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 18 Dec 2008 18:05:06 +0100
Subject: [Freeswitch-users] "choppy" voice
Message-ID: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>

Hello

I got problems with choppy voice. I just happens1 time of 10 or something
like that.
Incorrect call: http://pastebin.freeswitch.org/6479
working call: http://pastebin.freeswitch.org/6481

Any idea what is wrong?

Thanks,
  Jonas
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From odermann at googlemail.com  Thu Dec 18 09:07:20 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 18:07:20 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
Message-ID: <5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>

sorry, i do not know that. i could ask tomorrow.

is deflect, what i understand? the provider has to support it? if yes,
what could i tell and ask the provider, to find a solution to this
problem? the provider is quite open for new ideas, although we do not
want to be to dependant on the provider and his possibilities.


2008/12/18 Brian West <brian at freeswitch.org>:
> What switch is your provider using?
>
> /b
>
> On Dec 18, 2008, at 10:52 AM, Dennis wrote:
>
>> i had a look at the deflect app, but as far as i understand it, the
>> carrier has to support/understand it ans react on the signals.
>>
>> is that right or does this have nothing to do with our carrier? or
>> does this work between fs servers in the same local network?
>>
>>
>> another similar question is: how to reject calls, so that the carrier
>> tries to route the call to another fs? if we want to make changes to
>> fs and test them, we would like to block new incoming calls, till
>> there are no running calls, to shut down the fs.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Dec 18 09:19:19 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 11:19:19 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
	<5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
Message-ID: <6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>

Well its a standard SIP Refer, They may not support it for good reason.

/b

On Dec 18, 2008, at 11:07 AM, Dennis wrote:

> is deflect, what i understand? the provider has to support it? if yes,
> what could i tell and ask the provider, to find a solution to this
> problem? the provider is quite open for new ideas, although we do not
> want to be to dependant on the provider and his possibilities.



From anthony.minessale at gmail.com  Thu Dec 18 09:20:57 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Dec 2008 11:20:57 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
Message-ID: <191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>

It seems to be related to 20ms vs 30ms ptime.

What are the 2 devices and what rev of FS are you on?

There was more code added in the last few weeks to smooth out this
occurrence.

you can also opt to declare your codec prefs at 30ms

PCMU at 30i instead of just PCMU in the conf.



On Thu, Dec 18, 2008 at 11:05 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello
>
> I got problems with choppy voice. I just happens1 time of 10 or something
> like that.
> Incorrect call: http://pastebin.freeswitch.org/6479
> working call: http://pastebin.freeswitch.org/6481
>
> Any idea what is wrong?
>
> Thanks,
>   Jonas
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Thu Dec 18 09:27:15 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 18:27:15 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
	<5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
	<6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>
Message-ID: <5e414ed0812180927l1ec186efk8c8bdb74337896a@mail.gmail.com>

so if they do not suport it (which has to be seen), is there another
way to redirect a call from one fs to another without the provider?
like redirect from one fs to the other over the local lan?



2008/12/18 Brian West <brian at freeswitch.org>:
> Well its a standard SIP Refer, They may not support it for good reason.
>
> /b
>
> On Dec 18, 2008, at 11:07 AM, Dennis wrote:
>
>> is deflect, what i understand? the provider has to support it? if yes,
>> what could i tell and ask the provider, to find a solution to this
>> problem? the provider is quite open for new ideas, although we do not
>> want to be to dependant on the provider and his possibilities.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Dec 18 09:29:19 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 11:29:19 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180927l1ec186efk8c8bdb74337896a@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
	<5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
	<6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>
	<5e414ed0812180927l1ec186efk8c8bdb74337896a@mail.gmail.com>
Message-ID: <81964BF7-5EDA-47AC-A1E5-81D1046AD977@freeswitch.org>

do they follow a 302 redirect?  Because if the call isn't answered yet  
then you can do a redirect

/b

On Dec 18, 2008, at 11:27 AM, Dennis wrote:

> so if they do not suport it (which has to be seen), is there another
> way to redirect a call from one fs to another without the provider?
> like redirect from one fs to the other over the local lan?



From jonas.gauffin at gmail.com  Thu Dec 18 09:30:14 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 18 Dec 2008 18:30:14 +0100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
Message-ID: <d021d3be0812180930p5edcffewc828110d8dced7ea@mail.gmail.com>

Hello
I've checked out the latest trunk, the problem is still left.
Im using a  Linksys SPA8000 analogue telephone adapter as one device. The
other call comes through the sip gateway (from PSTN).

I'll try to specify 30ms.

Regards,
  Jonas

On Thu, Dec 18, 2008 at 6:20 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> It seems to be related to 20ms vs 30ms ptime.
>
> What are the 2 devices and what rev of FS are you on?
>
> There was more code added in the last few weeks to smooth out this
> occurrence.
>
> you can also opt to declare your codec prefs at 30ms
>
> PCMU at 30i instead of just PCMU in the conf.
>
>
>
> On Thu, Dec 18, 2008 at 11:05 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>
>> Hello
>>
>> I got problems with choppy voice. I just happens1 time of 10 or something
>> like that.
>> Incorrect call: http://pastebin.freeswitch.org/6479
>> working call: http://pastebin.freeswitch.org/6481
>>
>> Any idea what is wrong?
>>
>> Thanks,
>>   Jonas
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Thu Dec 18 09:32:47 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Dec 2008 18:32:47 +0100
Subject: [Freeswitch-users] mod_shout and mp3 formats
In-Reply-To: <49475FDE.7080108@gmx.net>
References: <49475FDE.7080108@gmx.net>
Message-ID: <494A893F.9080509@gmx.net>

Today I also tried playing a wav file with the "play" application and it
worked. However accessing the same file through shout:// didn't work
with freeswitch (with Totem it worked).
The point is that FS plays the file for several seconds, but I don't
hear any sound.
I also looked at the libraries according to the wiki wiich should be
iunstalled, and they are there: Configure does not show any warnings.

Nobody has a clue what may be the problem here?

Best regards
Peter



Peter P GMX schrieb:
> I try to play mp3 I generated through Cepstral TTs and which I encoded
> via lame.
> However they won't play, so my question is: Which mp3 formats are supported?
>
> I generate the wav files by the following
> /opt/swift/bin/swift -n Katrin -p
> audio/channels=1,cst/f0_shift=.8,speech/rate=120,audio/sampling-rate=8000,audio/deadair=2
> -o $wavefile $text
>
> Then I convert to mp3 by the following variations:
> lame 46.wav 46.mp3
> lame -s 32 46.wav 46.mp3
> lame --preset 128 46.wav 46.mp3
> lame --resample 44.1 --preset 128 46.wav 46.mp3
> lame --resample 32 --preset 128 46.wav 46.mp3
> lame --resample 44.1 46.wav 46.mp3
> lame --resample 44.1 -m s --preset 128 46.wav 46.mp3
> lame --resample 44.1 -m s  46.wav 46.mp3
> lame --resample 44.1 -m s -b 128 46.wav 46.mp3
> lame --resample 44.1 -m s -B 24 46.wav 46.mp3
> lame --preset voice -v -B 64 -a 46.wav 46.mp3
>
> None of them worked with the playback application
> (shout://localhost/tts/46.mp3). The sound files had a length of between
> 2 and 5 sec. 2 Times during various tries they played at least
> partially. But at the next try they didn't play again. However I have a
> prerecorded sound file (44.1KHz, 128 kBits stereo music) which always
> plays well.
> The console shows me that all files are successfully played and I get a
> channel_ececute and a channel_ececute_complete after some seconds during
> event_socket. But I don't hear any sound.
>
> All above samples however played well with Totem on Ubuntu.
>
> The wiki tells me that almost any mp3 format should play. What am I
> doing wrong here?
>
> Another question: Should normal wav files play as well? Also with wav I
> cannot hear any sound.
>
> Best regards
> Peter
>
>
>
>
>
>   


From odermann at googlemail.com  Thu Dec 18 09:41:47 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 18:41:47 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <81964BF7-5EDA-47AC-A1E5-81D1046AD977@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
	<5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
	<6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>
	<5e414ed0812180927l1ec186efk8c8bdb74337896a@mail.gmail.com>
	<81964BF7-5EDA-47AC-A1E5-81D1046AD977@freeswitch.org>
Message-ID: <5e414ed0812180941o4699286hc500a9d8d8d67ea9@mail.gmail.com>

sorry, this is to difficult for me. what does that mean?

they pass a call to one of our fs. then we see, that the call should
be on another fs. we know, that the call is on the wrong fs, before we
send an answer. so we could react accordingly.



2008/12/18 Brian West <brian at freeswitch.org>:
> do they follow a 302 redirect?  Because if the call isn't answered yet
> then you can do a redirect


From gkuri at ieee.org  Thu Dec 18 09:43:57 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 18 Dec 2008 09:43:57 -0800
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
Message-ID: <494A8BDD.1070007@ieee.org>

I've tried to do the same and in my own experience, most carriers don't
accept 302 redirects. What I've seen is they take the 302 as a failure
and move on to the next switch, so worse case with 3 switches, it will
take 2 retries before hitting the switch you want them to redirect to.

Gabe

Dennis wrote:
> i would like to know, what the best way is, to redirect an incoming
> call from one fs (fs1) to another fs (fs2).
> 
> we use 3 freeswitch servers and the carrier passes calls to the three
> fs servers randomly. if on fs server is not offline, the carrier sends
> the call to the next fs.
> this is generally good, but for conferencing it not so good.
> 
> i am using socket outbound and need to do this for conferencing. let's
> say, we have a conference going on on fs1. another person wants to
> enter this conference, but the call is passed to fs2. on fs2 we see,
> that the caller wants to enter the conference going on on fs1.
> 
> now we have to redirect the call from fs2 to fs1. is this done with
> "redirect" and some according settings/params or are there other ways
> to do this? we would like to do this without our carrier doing
> something, to be a little more independant.
> 
> thanks
> dennis
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mszlazak at aol.com  Thu Dec 18 09:45:24 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Thu, 18 Dec 2008 12:45:24 -0500
Subject: [Freeswitch-users] Crackling noise when bypassing media between
	endpoints.
Message-ID: <8CB2F4F8A23FF73-CC8-4D0@Webmail-mg18.sim.aol.com>

When using bypass_media (aka. no_media) mode between an X-lite softphone and Prophacy ASR, I get intermittent "crackiling" background noise with the audio that I'm hearing. 
How do I get rid of this?
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From odermann at googlemail.com  Thu Dec 18 09:58:21 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 18:58:21 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <494A8BDD.1070007@ieee.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org>
Message-ID: <5e414ed0812180958x23fc0836v89c9ddab742e6895@mail.gmail.com>

so at least they should react on a 302? this could help, although i do
not really understand, what happens on a 302.

if they support it, they would receive the target fs server ip where
they should try next with deflect?

if everything does not help and is not possible: what could i do else?
it would be very helpful, if fs would support another way, if the
provider does not offer specific features.



2008/12/18 Gabriel Kuri <gkuri at ieee.org>:
> I've tried to do the same and in my own experience, most carriers don't
> accept 302 redirects. What I've seen is they take the 302 as a failure
> and move on to the next switch, so worse case with 3 switches, it will
> take 2 retries before hitting the switch you want them to redirect to.
>
> Gabe
>
> Dennis wrote:
>> i would like to know, what the best way is, to redirect an incoming
>> call from one fs (fs1) to another fs (fs2).
>>
>> we use 3 freeswitch servers and the carrier passes calls to the three
>> fs servers randomly. if on fs server is not offline, the carrier sends
>> the call to the next fs.
>> this is generally good, but for conferencing it not so good.
>>
>> i am using socket outbound and need to do this for conferencing. let's
>> say, we have a conference going on on fs1. another person wants to
>> enter this conference, but the call is passed to fs2. on fs2 we see,
>> that the caller wants to enter the conference going on on fs1.
>>
>> now we have to redirect the call from fs2 to fs1. is this done with
>> "redirect" and some according settings/params or are there other ways
>> to do this? we would like to do this without our carrier doing
>> something, to be a little more independant.
>>
>> thanks
>> dennis
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From intralanman at freeswitch.org  Thu Dec 18 09:58:41 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 18 Dec 2008 12:58:41 -0500
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <494A8BDD.1070007@ieee.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org>
Message-ID: <494A8F51.3040003@freeswitch.org>

Gabriel Kuri wrote:
> I've tried to do the same and in my own experience, most carriers don't
> accept 302 redirects. What I've seen is they take the 302 as a failure
> and move on to the next switch, so worse case with 3 switches, it will
> take 2 retries before hitting the switch you want them to redirect to.
>
>   

could also just respond with a 503 in which case all carriers should 
fail over to the next one...

-Ray


From sicfslist at gmail.com  Thu Dec 18 10:04:08 2008
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 18 Dec 2008 12:04:08 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <494A8F51.3040003@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
Message-ID: <35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>

I agree with Ray ... using a 3XX series message is a bad idea ... or you
could put OpenSer in front using the LCR module ... 503 to OpenSer and it
would route to the next gateway in the gateway group.
I have yet to work with any carrier that handles 3XX series correctly except
for some of the TCAP guys.

On Thu, Dec 18, 2008 at 11:58 AM, Raymond Chandler <
intralanman at freeswitch.org> wrote:

> Gabriel Kuri wrote:
> > I've tried to do the same and in my own experience, most carriers don't
> > accept 302 redirects. What I've seen is they take the 302 as a failure
> > and move on to the next switch, so worse case with 3 switches, it will
> > take 2 retries before hitting the switch you want them to redirect to.
> >
> >
>
> could also just respond with a 503 in which case all carriers should
> fail over to the next one...
>
> -Ray
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From c_cav_01 at yahoo.com  Thu Dec 18 10:14:56 2008
From: c_cav_01 at yahoo.com (Chris)
Date: Thu, 18 Dec 2008 10:14:56 -0800 (PST)
Subject: [Freeswitch-users] Crackling noise when bypassing media between
	endpoints.
In-Reply-To: <8CB2F4F8A23FF73-CC8-4D0@Webmail-mg18.sim.aol.com>
Message-ID: <690698.862.qm@web55103.mail.re4.yahoo.com>

I'm no expert, but I believe in media bypass mode freeswitch isn't handling media so it's not a fs fix, it would be the quality of connection for each of the originator/terminator, fs just directs each endpoint to set's up a point to point connection for RTP.

Is this right?

mszlazak at aol.com wrote: When using bypass_media (aka. no_media) mode between an X-lite softphone and Prophacy ASR, I get intermittent "crackiling" background noise with the audio that I'm hearing. 
 How do I get rid of this?
 

---------------------------------
Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
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From c_cav_01 at yahoo.com  Thu Dec 18 10:18:36 2008
From: c_cav_01 at yahoo.com (Chris)
Date: Thu, 18 Dec 2008 10:18:36 -0800 (PST)
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180958x23fc0836v89c9ddab742e6895@mail.gmail.com>
Message-ID: <476921.94215.qm@web55103.mail.re4.yahoo.com>

Could you set up the 3 inbound call handlers, then set up a 4th switch with the "conference" domain, or if you don't want to set up a 4th, designate one of the 3 inbound switches with a "conference" domain to handle all conferences, do media bypass and bridge the calls to the "conference" domain?

Dennis <odermann at googlemail.com> wrote: so at least they should react on a 302? this could help, although i do
not really understand, what happens on a 302.

if they support it, they would receive the target fs server ip where
they should try next with deflect?

if everything does not help and is not possible: what could i do else?
it would be very helpful, if fs would support another way, if the
provider does not offer specific features.



2008/12/18 Gabriel Kuri :
> I've tried to do the same and in my own experience, most carriers don't
> accept 302 redirects. What I've seen is they take the 302 as a failure
> and move on to the next switch, so worse case with 3 switches, it will
> take 2 retries before hitting the switch you want them to redirect to.
>
> Gabe
>
> Dennis wrote:
>> i would like to know, what the best way is, to redirect an incoming
>> call from one fs (fs1) to another fs (fs2).
>>
>> we use 3 freeswitch servers and the carrier passes calls to the three
>> fs servers randomly. if on fs server is not offline, the carrier sends
>> the call to the next fs.
>> this is generally good, but for conferencing it not so good.
>>
>> i am using socket outbound and need to do this for conferencing. let's
>> say, we have a conference going on on fs1. another person wants to
>> enter this conference, but the call is passed to fs2. on fs2 we see,
>> that the caller wants to enter the conference going on on fs1.
>>
>> now we have to redirect the call from fs2 to fs1. is this done with
>> "redirect" and some according settings/params or are there other ways
>> to do this? we would like to do this without our carrier doing
>> something, to be a little more independant.
>>
>> thanks
>> dennis
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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From mike at jerris.com  Thu Dec 18 10:21:18 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Dec 2008 13:21:18 -0500
Subject: [Freeswitch-users] Gtalk to sip problems when reconfiguring
	from scratch
In-Reply-To: <op.umc43xs3atcbto@linux.site>
References: <op.umc43xs3atcbto@linux.site>
Message-ID: <E2C95942-F4AA-4AE6-8E97-DAFADE7967DD@jerris.com>

We do not support registration fetching.

Mike

On Dec 18, 2008, at 9:56 AM, kriko wrote:

> I recently purged all freeswitch config and restarted configuring from
> scratch. Using defaults, I modified public.xml dialplan config  
> (added line
> 16 - 28):
> http://pastebin.com/m5ece6e6f
>
> and added a new config under jingle_profiles:
> http://pastebin.com/d6e983b99
>
> I register with phonelite or twinkle as user 1000 (it says successfull
> reg.), but when I make a call from gtalk I hear that user is not
> available, none of
> client rings.
>
> When registering sofia prints:
> 2008-12-18 15:53:32 [INFO] sofia_presence.c:475
> actual_sofia_presence_event_handler() internal  
> START_PRESENCE_PROBE_SQL
> 2008-12-18 15:53:32 [NOTICE] sofia_presence.c:793
> sofia_presence_resub_callback() internal PRESENCE_PROBE 1000 at 10.99.8.221
> 2008-12-18 15:53:32 [INFO] sofia_presence.c:484
> actual_sofia_presence_event_handler() internal END_PRESENCE_PROBE_SQL
>
> 2008-12-18 15:53:32 [INFO] sofia_presence.c:547
> actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal)
> 2008-12-18 15:53:32 [INFO] sofia_presence.c:563
> actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal)
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:517
> actual_sofia_presence_event_handler() external is passive, skipping
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510
> actual_sofia_presence_event_handler() 10.99.8.221 is an alias,  
> skipping
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510
> actual_sofia_presence_event_handler() default is an alias, skipping
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510
> actual_sofia_presence_event_handler() nat is an alias, skipping
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510
> actual_sofia_presence_event_handler() outbound is an alias, skipping
>
> and when fetching registrations:
> 2008-12-18 15:53:50 [ERR] sofia_reg.c:1120
> sofia_reg_handle_sip_i_register() NO CONTACT!
>
> Does this means that registration failed? Why it doesn't call  
> anymore the
> targeted user (1000)?
>
> Here is also an extract from console log while calling, it caught my
> attention:
> 2008-12-18 15:54:41 [WARNING] mod_dptools.c:2047  
> user_outgoing_channel()
> Can't find user [1000@]
> 2008-12-18 15:54:41 [ERR] switch_ivr_originate.c:1110
> switch_ivr_originate() Cannot create outgoing channel of type [user]
> cause: [SUBSCRIBER_ABSENT]
> 2008-12-18 15:54:41 [DEBUG] switch_ivr_originate.c:1689
> switch_ivr_originate() Originate Resulted in Error Cause: 20
> [SUBSCRIBER_ABSENT]
>


From c_cav_01 at yahoo.com  Thu Dec 18 10:28:00 2008
From: c_cav_01 at yahoo.com (Chris)
Date: Thu, 18 Dec 2008 10:28:00 -0800 (PST)
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <476921.94215.qm@web55103.mail.re4.yahoo.com>
Message-ID: <365310.11415.qm@web55103.mail.re4.yahoo.com>

If you need to do load balancing, you could set up a conference_a domain on one switch, conference_b on the second, conference_c on the third, then use xml_curl to dialplan and bridge the call to the right domain...

But again, I am no expert...  Just a noob trying to be creative.  :P

Chris <c_cav_01 at yahoo.com> wrote: Could you set up the 3 inbound call handlers, then set up a 4th switch with the "conference" domain, or if you don't want to set up a 4th, designate one of the 3 inbound switches with a "conference" domain to handle all conferences, do media bypass and bridge the calls to the "conference" domain?

Dennis <odermann at googlemail.com> wrote: so at least they should react on a 302? this could help, although i do
not really understand, what happens on a 302.

if they support it, they would receive the target fs server ip where
they should try next with deflect?

if everything does not help and is not possible: what could i do else?
it would be very helpful, if fs would support another way, if the
provider does not offer specific features.



2008/12/18 Gabriel Kuri :
> I've tried to do  the same and in my own experience, most carriers don't
> accept 302 redirects. What I've seen is they take the 302 as a failure
> and move on to the next switch, so worse case with 3 switches, it will
> take 2 retries before hitting the switch you want them to redirect to.
>
> Gabe
>
> Dennis wrote:
>> i would like to know, what the best way is, to redirect an incoming
>> call from one fs (fs1) to another fs (fs2).
>>
>> we use 3 freeswitch servers and the carrier passes calls to the three
>> fs servers randomly. if on fs server is not offline, the carrier sends
>> the call to the next fs.
>> this is generally good, but for conferencing it not so good.
>>
>> i am using socket outbound and need to do this for conferencing. let's
>> say, we have a conference going on on fs1. another person wants to
>> enter this conference, but the call  is passed to fs2. on fs2 we see,
>> that the caller wants to enter the conference going on on fs1.
>>
>> now we have to redirect the call from fs2 to fs1. is this done with
>> "redirect" and some according settings/params or are there other ways
>> to do this? we would like to do this without our carrier doing
>> something, to be a little more independant.
>>
>> thanks
>> dennis
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From Prometheus001 at gmx.net  Thu Dec 18 10:29:11 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Dec 2008 19:29:11 +0100
Subject: [Freeswitch-users] mod_shout and mp3 formats
In-Reply-To: <49475FDE.7080108@gmx.net>
References: <49475FDE.7080108@gmx.net>
Message-ID: <494A9677.8030608@gmx.net>

Today I also tried playing a wav file with the "play" application and it
worked. However accessing the same file through shout:// didn't work
with freeswitch (with Totem it worked).
The point is that FS plays the file for several seconds, but I don't
hear any sound.
I also looked at the libraries according to the wiki wiich should be
iunstalled, and they are there: Configure does not show any warnings.

Nobody has a clue what may be the problem here?

Best regards
Peter



Peter P GMX schrieb:
> I try to play mp3 I generated through Cepstral TTs and which I encoded
> via lame.
> However they won't play, so my question is: Which mp3 formats are supported?
>
> I generate the wav files by the following
> /opt/swift/bin/swift -n Katrin -p
> audio/channels=1,cst/f0_shift=.8,speech/rate=120,audio/sampling-rate=8000,audio/deadair=2
> -o $wavefile $text
>
> Then I convert to mp3 by the following variations:
> lame 46.wav 46.mp3
> lame -s 32 46.wav 46.mp3
> lame --preset 128 46.wav 46.mp3
> lame --resample 44.1 --preset 128 46.wav 46.mp3
> lame --resample 32 --preset 128 46.wav 46.mp3
> lame --resample 44.1 46.wav 46.mp3
> lame --resample 44.1 -m s --preset 128 46.wav 46.mp3
> lame --resample 44.1 -m s  46.wav 46.mp3
> lame --resample 44.1 -m s -b 128 46.wav 46.mp3
> lame --resample 44.1 -m s -B 24 46.wav 46.mp3
> lame --preset voice -v -B 64 -a 46.wav 46.mp3
>
> None of them worked with the playback application
> (shout://localhost/tts/46.mp3). The sound files had a length of between
> 2 and 5 sec. 2 Times during various tries they played at least
> partially. But at the next try they didn't play again. However I have a
> prerecorded sound file (44.1KHz, 128 kBits stereo music) which always
> plays well.
> The console shows me that all files are successfully played and I get a
> channel_ececute and a channel_ececute_complete after some seconds during
> event_socket. But I don't hear any sound.
>
> All above samples however played well with Totem on Ubuntu.
>
> The wiki tells me that almost any mp3 format should play. What am I
> doing wrong here?
>
> Another question: Should normal wav files play as well? Also with wav I
> cannot hear any sound.
>
> Best regards
> Peter
>
>
>
>
>
>   



From brian at freeswitch.org  Thu Dec 18 10:31:11 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 12:31:11 -0600
Subject: [Freeswitch-users] Crackling noise when bypassing media between
	endpoints.
In-Reply-To: <690698.862.qm@web55103.mail.re4.yahoo.com>
References: <690698.862.qm@web55103.mail.re4.yahoo.com>
Message-ID: <22787EB4-371A-4907-8C9D-96D7F2782AF2@freeswitch.org>

Yes Chris you are right.  FreeSWITCH isn't involved in the media at all.

/b

On Dec 18, 2008, at 12:14 PM, Chris wrote:

> I'm no expert, but I believe in media bypass mode freeswitch isn't  
> handling media so it's not a fs fix, it would be the quality of  
> connection for each of the originator/terminator, fs just directs  
> each endpoint to set's up a point to point connection for RTP.
>
> Is this right?



From odermann at googlemail.com  Thu Dec 18 10:36:04 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 19:36:04 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <494A8F51.3040003@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
Message-ID: <5e414ed0812181036q297d7f82vcec9b8b0e3cf8cee@mail.gmail.com>

thanks for all your help!

this sounds interesting. it seems, that these codes should be
available by default with sip!? is this right?

i will talk to the carrier tomorrow and ask, what is possible.

as far as i can see, i am always dependant on the carrier? there is no
way to pass a call from one fs to another?




2008/12/18 Raymond Chandler <intralanman at freeswitch.org>:
> Gabriel Kuri wrote:
>> I've tried to do the same and in my own experience, most carriers don't
>> accept 302 redirects. What I've seen is they take the 302 as a failure
>> and move on to the next switch, so worse case with 3 switches, it will
>> take 2 retries before hitting the switch you want them to redirect to.
>>
>>
>
> could also just respond with a 503 in which case all carriers should
> fail over to the next one...
>
> -Ray
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Thu Dec 18 10:40:03 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Dec 2008 13:40:03 -0500
Subject: [Freeswitch-users] mod_shout and mp3 formats
In-Reply-To: <494A9677.8030608@gmx.net>
References: <49475FDE.7080108@gmx.net> <494A9677.8030608@gmx.net>
Message-ID: <04209D71-419B-4346-9A9F-C9D85B566A8F@jerris.com>

shout does not play wav files it plays mp3 files.

Mike

On Dec 18, 2008, at 1:29 PM, Peter P GMX wrote:

> Today I also tried playing a wav file with the "play" application  
> and it
> worked. However accessing the same file through shout:// didn't work
> with freeswitch (with Totem it worked).
> The point is that FS plays the file for several seconds, but I don't
> hear any sound.
> I also looked at the libraries according to the wiki wiich should be
> iunstalled, and they are there: Configure does not show any warnings.
>
> Nobody has a clue what may be the problem here?
>
> Best regards
> Peter



From mszlazak at aol.com  Thu Dec 18 10:49:02 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Thu, 18 Dec 2008 13:49:02 -0500
Subject: [Freeswitch-users] Crackling noise when bypassing media between
 endpoints.
In-Reply-To: <22787EB4-371A-4907-8C9D-96D7F2782AF2@freeswitch.org>
References: <690698.862.qm@web55103.mail.re4.yahoo.com>
	<22787EB4-371A-4907-8C9D-96D7F2782AF2@freeswitch.org>
Message-ID: <8CB2F586D303ED3-CC8-8B6@Webmail-mg18.sim.aol.com>


 Man, I can't win with this one. 

I can bypass media between two endpoints with some "static" but what I really want FS to do is process the audio before it's passed on.
However, getting FS involved is something I haven't had any success in with these two endpoints ... so far.

Thanks for pointing out the noise source(s) with bypass ... makes sense given the name.


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, 18 Dec 2008 10:31 am
Subject: Re: [Freeswitch-users] Crackling noise when bypassing media between endpoints.










Yes Chris you are right.  FreeSWITCH isn't involved in the media at all.

/b

On Dec 18, 2008, at 12:14 PM, Chris wrote:

> I'm no expert, but I believe in media bypass mode freeswitch isn't  
> handling media so it's not a fs fix, it would be the quality of  
> connection for each of the originator/terminator, fs just directs  
> each endpoint to set's up a point to point connection for RTP.
>
> Is this right?


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Thu Dec 18 11:00:26 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Dec 2008 13:00:26 -0600
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <494A6DE0.4030701@networkoblivion.com>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<494A6DE0.4030701@networkoblivion.com>
Message-ID: <191c3a030812181100jd7a0c20m97596121650d404b@mail.gmail.com>

Can you answer the questions and possibly go online on IRC so we can debug
your issue?


On Thu, Dec 18, 2008 at 9:36 AM, peder at networkoblivion.com <
peder at networkoblivion.com> wrote:

> If anybody wants to look at the core dump in gdb, here it is (the actual
> core is 256Meg):
>
> http://pastebin.freeswitch.org/6476
>
> I know zip about debugging and gdb, but from looking through it, I see a
> segmentation fault and it appears to be thread 15094.  The last three
> items in the bt full for that thread are:
>
> destroy_status = <value optimized out>
> fd = (switch_file_t *) 0x80529b0
> pool = (switch_memory_pool_t *) 0x80528f0
>
> So I would guess it is trying to access an invalid memory location, but
> why, I have no idea....
>
> Any ideas?
>
>
> Peder
>
>
> Michael S Collins wrote:
> > Check out this page:
> > wiki.freeswitch.org/wiki/Debugging_Freeswitch
> >
> > -MC
> >
> > Sent from my iPhone
> >
> > On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <
> peder at networkoblivion.com
> >  > wrote:
> >
> >> What is the process for capturing and submitting a core dump?
> >>
> >> I am messing around with the Cisco 79x1 phones and tcp and multiple
> >> reg.
> >> I have a 7961 using tcp and a 7960 using udp both reg'd with the same
> >> number and both showing up as registered.  If I call out from the
> >> phone
> >> using tcp, it works.  If I call out from the phone using udp, I get a
> >> core dump.  If I call in, it calls both phones and I am assuming the
> >> call to the phone using udp causes a core dump as well.  These are the
> >> only two phones on the system and I am running version 10851 from
> >> yesterday.  If I only have the udp phone registered and the tcp
> >> phone is
> >> off, it works fine.  It is only when I have a mix of a udp reg and a
> >> tcp
> >> reg on the same number that I appear to get a core dump.
> >>
> >>
> >> Peder
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From intralanman at freeswitch.org  Thu Dec 18 11:05:06 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 18 Dec 2008 14:05:06 -0500
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21069519.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com>
	<49490301.8040701@lists.rupa.com>	<21054080.post@talk.nabble.com>
	<49491847.3030403@lists.rupa.com> <21069519.post@talk.nabble.com>
Message-ID: <494A9EE2.7050507@freeswitch.org>

Carole O. wrote:
> Hello,
>
> Thanks for your answers!
> Concerning the creation of a new variable for the conference the problem is
> that I do not create channels from the conference. I call separately a new
> member on a new channel and add it on the conference only if he agrees to
> enter it. So it was the same problem as for the uuid, I am not sure I can
> access the good variable from anywhere in case many conferences are running.
>
>   
you could use the db app to hold state across multiple calls... maybe 
use the ${caller_id_number} and the ${destination_number} as keys for 
the insert/select so that there's something constant to use in the 
select... and another extension or two may be needed...

You could do the db lookup before you make the call so that you see if 
your caller is already a member of a conference.... if he is, then the 
transfer from *1 would work much the same as it does now except you'd 
use the result of the db lookup as the conference number... if he's not 
a member of an existing conference, then you could generate the uuid 
like Anthony said before, then do a db insert for ${caller_id_number} 
and ${destination_number} to insert that newly created uuid and use it 
as the conference number.... one caveat that i see here is that the 
destination_number would have to be exactly the same as if that user 
were callling and it was his caller_id_number, otherwise your query will 
fail.

you'll also need to "clean" the db when you hangup, which should be able 
to be accomplished with an execute_on_hangup that does a delete of the 
conf data for each user

-Ray


From kkielhofner at star2star.com  Thu Dec 18 11:21:49 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Thu, 18 Dec 2008 14:21:49 -0500
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
	<35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>
Message-ID: <2d9149cd0812181121v47823968u6fb8f40aa86c5fb3@mail.gmail.com>

On Thu, Dec 18, 2008 at 1:04 PM, Shelby Ramsey <sicfslist at gmail.com> wrote:
> I agree with Ray ... using a 3XX series message is a bad idea ... or you
> could put OpenSer in front using the LCR module ... 503 to OpenSer and it
> would route to the next gateway in the gateway group.
> I have yet to work with any carrier that handles 3XX series correctly except
> for some of the TCAP guys.
>

Level(3) readily supports 302s if the destination IP of the new
contact has been made known to Level(3) beforehand.  You can't 302
just anywhere but you can 302 to your own boxes, networks, etc.



-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From brian at freeswitch.org  Thu Dec 18 11:27:53 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 13:27:53 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <2d9149cd0812181121v47823968u6fb8f40aa86c5fb3@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
	<35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>
	<2d9149cd0812181121v47823968u6fb8f40aa86c5fb3@mail.gmail.com>
Message-ID: <A8991787-3B4E-480C-A137-685C5884BD8E@freeswitch.org>

Excellent advice.   So just letting L3 know the IP's and you'll be fine.

/b

On Dec 18, 2008, at 1:21 PM, Kristian Kielhofner wrote:

>
> Level(3) readily supports 302s if the destination IP of the new
> contact has been made known to Level(3) beforehand.  You can't 302
> just anywhere but you can 302 to your own boxes, networks, etc.



From peder at networkoblivion.com  Thu Dec 18 11:33:31 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Thu, 18 Dec 2008 13:33:31 -0600
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <CAEDC248-D670-4603-A4A8-5F06074304E9@freeswitch.org>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>	<494A6DE0.4030701@networkoblivion.com>
	<CAEDC248-D670-4603-A4A8-5F06074304E9@freeswitch.org>
Message-ID: <494AA58B.9030500@networkoblivion.com>

I can make it happen on demand.  All I have to do is call the shared 
number and it crashes.  I'll hop on IRC in a bit.

Michael S Collins wrote:
> Is this a single occurrence or can you make it happen consistently?
> -MC
> 
> Sent from my iPhone
> 
> On Dec 18, 2008, at 7:36 AM, "peder at networkoblivion.com" <peder at networkoblivion.com 
>  > wrote:
> 
>> If anybody wants to look at the core dump in gdb, here it is (the  
>> actual
>> core is 256Meg):
>>
>> http://pastebin.freeswitch.org/6476
>>
>> I know zip about debugging and gdb, but from looking through it, I  
>> see a
>> segmentation fault and it appears to be thread 15094.  The last three
>> items in the bt full for that thread are:
>>
>> destroy_status = <value optimized out>
>> fd = (switch_file_t *) 0x80529b0
>> pool = (switch_memory_pool_t *) 0x80528f0
>>
>> So I would guess it is trying to access an invalid memory location,  
>> but
>> why, I have no idea....
>>
>> Any ideas?
>>
>>
>> Peder
>>
>>
>> Michael S Collins wrote:
>>> Check out this page:
>>> wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>
>>> -MC
>>>
>>> Sent from my iPhone
>>>
>>> On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com
>>>> wrote:
>>>> What is the process for capturing and submitting a core dump?
>>>>
>>>> I am messing around with the Cisco 79x1 phones and tcp and multiple
>>>> reg.
>>>> I have a 7961 using tcp and a 7960 using udp both reg'd with the  
>>>> same
>>>> number and both showing up as registered.  If I call out from the
>>>> phone
>>>> using tcp, it works.  If I call out from the phone using udp, I  
>>>> get a
>>>> core dump.  If I call in, it calls both phones and I am assuming the
>>>> call to the phone using udp causes a core dump as well.  These are  
>>>> the
>>>> only two phones on the system and I am running version 10851 from
>>>> yesterday.  If I only have the udp phone registered and the tcp
>>>> phone is
>>>> off, it works fine.  It is only when I have a mix of a udp reg and a
>>>> tcp
>>>> reg on the same number that I appear to get a core dump.
>>>>
>>>>
>>>> Peder
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From scott.ellis at novatex.com.au  Thu Dec 18 15:04:54 2008
From: scott.ellis at novatex.com.au (Scott Ellis)
Date: Fri, 19 Dec 2008 10:04:54 +1100
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
In-Reply-To: <191c3a030812180612n94f9b72vc9ad0ba2d90d6a9e@mail.gmail.com>
References: <49478B0F.3000802@novatex.com.au>	<153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>	<4949B946.5050502@novatex.com.au>
	<191c3a030812180612n94f9b72vc9ad0ba2d90d6a9e@mail.gmail.com>
Message-ID: <494AD716.1010708@novatex.com.au>

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From jason at jasonjgw.net  Thu Dec 18 20:35:09 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Dec 2008 15:35:09 +1100
Subject: [Freeswitch-users] debug symbols (was Re: Core Dump)
In-Reply-To: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
Message-ID: <20081219043509.GA4225@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 07:02:18AM -0800, Michael S Collins wrote:
> Check out this page:
> wiki.freeswitch.org/wiki/Debugging_Freeswitch

In the long term (i.e., when more important matters aren't at issue), it might
be a good idea to modify the build process so that the debug symbols are
written out as separate files. In recent Linux distributions such as Debian
and Fedora, the debug symbols are often provided in separate packages that can
be installed whenever needed.



From marc at kasteris.com  Thu Dec 18 21:46:39 2008
From: marc at kasteris.com (Marc Orenberg)
Date: Thu, 18 Dec 2008 21:46:39 -0800 (PST)
Subject: [Freeswitch-users] Ending a bridged call with a touchtone
References: <mailman.2512.1229488183.18013.freeswitch-users@lists.freeswitch.org>
Message-ID: <876517.26535.qm@web50805.mail.re2.yahoo.com>

Thanks for the response Brian.? I don't understand what bind_meta does, or how it can help me. Is it something I can use from my Python script?? I searched for a description of it, but I was unable to find one.? Could you please point me towards some documentation, or maybe quickly explain it?? Thanks.

?
>Date: Tue, 16 Dec 2008 11:30:46 -0600
>From: Brian West <brian at freeswitch.org>
>Subject: Re: [Freeswitch-users] Ending a bridged call with a touchtone
>To: freeswitch-users at lists.freeswitch.org
>Message-ID: <F9E12702-5CA0-45A8-B408-3F2162F94A8E at freeswitch.org>
>Content-Type: text/plain; charset="us-ascii"
>
>Try bind_meta, examples are in the default dialplan.
>
>/b
>
>On Dec 16, 2008, at 10:24 PM, Marc Orenberg wrote:
>
>> Hello.? I'm trying to allow the A-leg of a bridged call to be able? 
>> to press a touchtone to end the call.
>> In my Python script, I set-up a DTMF callback function using? 
>> setInputCallback, but it doesn't seem to have any effect during? 
>> bridged calls. Is there another way to do this? Thanks.
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From mike at jerris.com  Thu Dec 18 21:24:24 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Dec 2008 00:24:24 -0500
Subject: [Freeswitch-users] debug symbols (was Re: Core Dump)
In-Reply-To: <20081219043509.GA4225@jdc.jasonjgw.net>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<20081219043509.GA4225@jdc.jasonjgw.net>
Message-ID: <02B71311-5C31-4B19-BD62-6DBC9958F774@jerris.com>

This is a function of the packaging system, not the build system and  
at least the debs do have this already.  On a related note I fixed the  
Sofia build to include proper symbols now all the time in the debug  
build.

Mike

On Dec 18, 2008, at 11:35 PM, Jason White <jason at jasonjgw.net> wrote:

> On Thu, Dec 18, 2008 at 07:02:18AM -0800, Michael S Collins wrote:
>> Check out this page:
>> wiki.freeswitch.org/wiki/Debugging_Freeswitch
>
> In the long term (i.e., when more important matters aren't at  
> issue), it might
> be a good idea to modify the build process so that the debug symbols  
> are
> written out as separate files. In recent Linux distributions such as  
> Debian
> and Fedora, the debug symbols are often provided in separate  
> packages that can
> be installed whenever needed.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jason at jasonjgw.net  Thu Dec 18 23:41:37 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Dec 2008 18:41:37 +1100
Subject: [Freeswitch-users] debug symbols (was Re: Core Dump)
In-Reply-To: <02B71311-5C31-4B19-BD62-6DBC9958F774@jerris.com>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<20081219043509.GA4225@jdc.jasonjgw.net>
	<02B71311-5C31-4B19-BD62-6DBC9958F774@jerris.com>
Message-ID: <20081219074137.GA6310@jdc.jasonjgw.net>

On Fri, Dec 19, 2008 at 12:24:24AM -0500, Michael Jerris wrote:
> This is a function of the packaging system, not the build system and  
> at least the debs do have this already.  On a related note I fixed the  
> Sofia build to include proper symbols now all the time in the debug  
> build.

Thanks, this is excellent.

The supportive community associated with this project is gratefully
appreciated.



From msc at freeswitch.org  Thu Dec 18 23:49:35 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Dec 2008 23:49:35 -0800
Subject: [Freeswitch-users] debug symbols (was Re: Core Dump)
In-Reply-To: <20081219074137.GA6310@jdc.jasonjgw.net>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<20081219043509.GA4225@jdc.jasonjgw.net>
	<02B71311-5C31-4B19-BD62-6DBC9958F774@jerris.com>
	<20081219074137.GA6310@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812182349kf65c76cr6f010e32e4ed4cd6@mail.gmail.com>

On Thu, Dec 18, 2008 at 11:41 PM, Jason White <jason at jasonjgw.net> wrote:

> On Fri, Dec 19, 2008 at 12:24:24AM -0500, Michael Jerris wrote:
> > This is a function of the packaging system, not the build system and
> > at least the debs do have this already.  On a related note I fixed the
> > Sofia build to include proper symbols now all the time in the debug
> > build.
>
> Thanks, this is excellent.
>
> The supportive community associated with this project is gratefully
> appreciated.


The members of the supportive community appreciate your appreciation! ;)
-MC


>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mszlazak at aol.com  Fri Dec 19 00:03:31 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Fri, 19 Dec 2008 03:03:31 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
Message-ID: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>

I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.
I'm trying to pass audio "internally" on the same machine between endpoints and have be advised that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 
However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.
Any comments?? 
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From fidibus83 at aol.com  Fri Dec 19 01:03:32 2008
From: fidibus83 at aol.com (fidibus83)
Date: Fri, 19 Dec 2008 10:03:32 +0100
Subject: [Freeswitch-users] Problem with openzap
In-Reply-To: <004801c960f1$dc680840$6445310a@Franzi>
References: <004801c960f1$dc680840$6445310a@Franzi>
Message-ID: <005e01c961b8$aba3b840$6445310a@Franzi>

Hello,

 

Here is the newbie in FS! I need your help again!

 

When FS is running I get every few seconds this warning:

 

[WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6 (or 7 or 8)

 

Why?

 

Do you need some configurations?

 

Thanks!

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From kristjan.ugrin at gmail.com  Fri Dec 19 01:09:49 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Fri, 19 Dec 2008 10:09:49 +0100
Subject: [Freeswitch-users] Java example
In-Reply-To: <191c3a030812180619k394a9e33g7cba5808e7d95558@mail.gmail.com>
References: <op.umcs7ncwatcbto@linux.site>
	<6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>
	<op.umcxenzsatcbto@linux.site>
	<191c3a030812180619k394a9e33g7cba5808e7d95558@mail.gmail.com>
Message-ID: <op.umejqlg7atcbto@linux.site>

Seems like my dialplan was a bit problematic, it works now.
Thanks.

On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale  
<anthony.minessale at gmail.com> wrote:

> did you turn up your console log level high enough to see it? The default
> level is "INFO"
>
>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>



-- 
kriko


From kristjan.ugrin at gmail.com  Fri Dec 19 01:12:49 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Fri, 19 Dec 2008 10:12:49 +0100
Subject: [Freeswitch-users] [Java] Catching dingaling messages
Message-ID: <op.umejvnuiatcbto@linux.site>

Hello!

I was wondering if it would be possible to catch messages from dingaling.  
I saw it can print out messages into console when a user types in a  
message,
but it doesn't understand it. I would like to catch that and do something  
(like initiate a call).
I know you have to call you program via a dialplan, so I don't know how  
would it be really possible to invoke your java app
when other events occur.

Is it possible?

Cheers,

-- 
kriko


From jason at jasonjgw.net  Fri Dec 19 01:22:07 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Dec 2008 20:22:07 +1100
Subject: [Freeswitch-users] [Java] Catching dingaling messages
In-Reply-To: <op.umejvnuiatcbto@linux.site>
References: <op.umejvnuiatcbto@linux.site>
Message-ID: <20081219092207.GA6967@jdc.jasonjgw.net>

On Fri, Dec 19, 2008 at 10:12:49AM +0100, kriko wrote:
> I was wondering if it would be possible to catch messages from dingaling.  
> I saw it can print out messages into console when a user types in a  
> message,
> but it doesn't understand it. I would like to catch that and do something  
> (like initiate a call).
> I know you have to call you program via a dialplan, so I don't know how  
> would it be really possible to invoke your java app
> when other events occur.

Listen to the event socket on port 8021 and register to receive the events you
want to monitor, then issue api commands via the same socket interface to make
the outbound call.

See the wiki page about the event socket.

I haven't looked at this in any detail, but it's clear from the wiki that this
should be able to meet your needs.



From jason at jasonjgw.net  Fri Dec 19 02:02:58 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Dec 2008 21:02:58 +1100
Subject: [Freeswitch-users] Ending a bridged call with a touchtone
In-Reply-To: <876517.26535.qm@web50805.mail.re2.yahoo.com>
References: <mailman.2512.1229488183.18013.freeswitch-users@lists.freeswitch.org>
	<876517.26535.qm@web50805.mail.re2.yahoo.com>
Message-ID: <20081219100258.GA7206@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 09:46:39PM -0800, Marc Orenberg wrote:
> Thanks for the response Brian.? I don't understand what bind_meta does, or
> how it can help me. Is it something I can use from my Python script?? I
> searched for a description of it, but I was unable to find one.? Could you
> please point me towards some documentation, or maybe quickly explain it??

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app



From fidibus83 at aol.com  Fri Dec 19 02:10:29 2008
From: fidibus83 at aol.com (fidibus83)
Date: Fri, 19 Dec 2008 11:10:29 +0100
Subject: [Freeswitch-users] Problem with openzap
In-Reply-To: <005e01c961b8$aba3b840$6445310a@Franzi>
References: <004801c960f1$dc680840$6445310a@Franzi>
	<005e01c961b8$aba3b840$6445310a@Franzi>
Message-ID: <006d01c961c2$06085990$6445310a@Franzi>

Hello,

 

I get more warnings yet:

 

[WARNING]  mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for
channel 1:1( it?s going to 1:31)

[WARNING] zap_isdn.c:803 process_event() channel 1:1 (1:2) (it?s going to
1:31 (1:16))

 

I don?t know what to do? Can you help me?

 

I have a Linux-Server with a Digium Wildcard TE110P.

 

Oz list:

 

API CALL [oz(list)] output:

+OK

span: 1

type: isdn

chan_count: 31

Dialplan: XML

context: default

dial_regex:

fial_dial_regex:

hold_music:

analog_options none

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
fidibus83
Gesendet: Freitag, 19. Dezember 2008 10:04
An: freeswitch-users at lists.freeswitch.org
Betreff: [Freeswitch-users] Problem with openzap

 

Hello,

 

Here is the newbie in FS! I need your help again!

 

When FS is running I get every few seconds this warning:

 

[WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6 (or 7 or 8)

 

Why?

 

Do you need some configurations?

 

Thanks!
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From odermann at googlemail.com  Fri Dec 19 04:18:52 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 19 Dec 2008 13:18:52 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <A8991787-3B4E-480C-A137-685C5884BD8E@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
	<35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>
	<2d9149cd0812181121v47823968u6fb8f40aa86c5fb3@mail.gmail.com>
	<A8991787-3B4E-480C-A137-685C5884BD8E@freeswitch.org>
Message-ID: <5e414ed0812190418j19c259e4m674c55fde7765de2@mail.gmail.com>

sendmsg redirect to an ip-adress of one of our fs server works great.
thanks for your help.

dannis


From Claudio.Cavalera at italtel.it  Fri Dec 19 04:36:45 2008
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Fri, 19 Dec 2008 13:36:45 +0100
Subject: [Freeswitch-users] Problem compiling socket2me
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0969642@BESONE.corp.dom>

Hello guys,
I'm playing with fs fax capabilities following these guidelines:
http://wiki.freeswitch.org/wiki/Examples_faxlib.jm

I've compiled mod_fax
with make mod_fax-install
and that should have taken care also of spandsp.

When I issue make in scripts/socket2me I get this error:
socket2me.c:315: error: 'fax_state_t' has no member named 't30_state'

Maybe something has changed in the api?
Could you please help me track down the problem?

Thanks,
Claudio


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From kristjan.ugrin at gmail.com  Fri Dec 19 07:37:52 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Fri, 19 Dec 2008 16:37:52 +0100
Subject: [Freeswitch-users] [Java] Catching dingaling messages
In-Reply-To: <20081219092207.GA6967@jdc.jasonjgw.net>
References: <op.umejvnuiatcbto@linux.site>
	<20081219092207.GA6967@jdc.jasonjgw.net>
Message-ID: <op.ume1pdtjatcbto@linux.site>

That's seems the right this, thanks.

But the dingaling is only returning this events:
  dingaling::login_success
  dingaling::login_failure
  dingaling::connected

Is it possible in any way to catch text messages?


On Fri, 19 Dec 2008 10:22:07 +0100, Jason White <jason at jasonjgw.net> wrote:

> On Fri, Dec 19, 2008 at 10:12:49AM +0100, kriko wrote:
>> I was wondering if it would be possible to catch messages from  
>> dingaling.
>> I saw it can print out messages into console when a user types in a
>> message,
>> but it doesn't understand it. I would like to catch that and do  
>> something
>> (like initiate a call).
>> I know you have to call you program via a dialplan, so I don't know how
>> would it be really possible to invoke your java app
>> when other events occur.
>
> Listen to the event socket on port 8021 and register to receive the  
> events you
> want to monitor, then issue api commands via the same socket interface  
> to make
> the outbound call.
>
> See the wiki page about the event socket.
>
> I haven't looked at this in any detail, but it's clear from the wiki  
> that this
> should be able to meet your needs.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Porn - the reason you need a new hard drive.


From msc at freeswitch.org  Fri Dec 19 07:45:04 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Dec 2008 07:45:04 -0800
Subject: [Freeswitch-users] Problem with openzap
In-Reply-To: <006d01c961c2$06085990$6445310a@Franzi>
References: <004801c960f1$dc680840$6445310a@Franzi>
	<005e01c961b8$aba3b840$6445310a@Franzi>
	<006d01c961c2$06085990$6445310a@Franzi>
Message-ID: <87f2f3b90812190745i2101335fw483990419659603e@mail.gmail.com>

Which dialect are you running and what is on the other end of the PRI?
-MC

On Fri, Dec 19, 2008 at 2:10 AM, fidibus83 <fidibus83 at aol.com> wrote:

>  Hello,
>
>
>
> I get more warnings yet:
>
>
>
> [WARNING]  mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for
> channel 1:1( it's going to 1:31)
>
> [WARNING] zap_isdn.c:803 process_event() channel 1:1 (1:2) (it's going to
> 1:31 (1:16))
>
>
>
> I don't know what to do? Can you help me?
>
>
>
> I have a Linux-Server with a Digium Wildcard TE110P.
>
>
>
> Oz list:
>
>
>
> API CALL [oz(list)] output:
>
> +OK
>
> span: 1
>
> type: isdn
>
> chan_count: 31
>
> Dialplan: XML
>
> context: default
>
> dial_regex:
>
> fial_dial_regex:
>
> hold_music:
>
> analog_options none
>
>
>  ------------------------------
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *fidibus83
> *Gesendet:* Freitag, 19. Dezember 2008 10:04
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* [Freeswitch-users] Problem with openzap
>
>
>
> Hello,
>
>
>
> Here is the newbie in FS! I need your help again!
>
>
>
> When FS is running I get every few seconds this warning:
>
>
>
> [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6 (or 7 or 8)
>
>
>
> Why?
>
>
>
> Do you need some configurations?
>
>
>
> Thanks!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Fri Dec 19 07:49:33 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Dec 2008 10:49:33 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
Message-ID: <B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>

It gives me the impression there is something wrong with your firewall  
running on the box.

Mike

On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:

> I find it strange that I can have to endpoints get audio went using  
> bypass media mode but the audio fails to go across endpoints if I  
> use proxy media mode.
> I'm trying to pass audio "internally" on the same machine between  
> endpoints and have be advised that a reason the audio may fail to be  
> passed is because there is some RTP timing and IP address/port issues.
> However, FS has no problem "connecting" ports if i change the mode  
> to bypass media. This gives me the impression that something is  
> wrong with FS proxy media mode.
> Any comments?
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Dec 19 07:52:57 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Dec 2008 10:52:57 -0500
Subject: [Freeswitch-users] Problem compiling socket2me
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0969642@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0969642@BESONE.corp.dom>
Message-ID: <94326D53-5AFD-4B71-845D-41F486D02D10@jerris.com>

mod_fax replaces socket2me, you don't need it anymore.

Mike

On Dec 19, 2008, at 7:36 AM, Cavalera Claudio Luigi wrote:

> Hello guys,
> I'm playing with fs fax capabilities following these guidelines:
> http://wiki.freeswitch.org/wiki/Examples_faxlib.jm
>
> I've compiled mod_fax
> with make mod_fax-install
> and that should have taken care also of spandsp.
>
> When I issue make in scripts/socket2me I get this error:
> socket2me.c:315: error: 'fax_state_t' has no member named 't30_state'
>
> Maybe something has changed in the api?
> Could you please help me track down the problem?
>
> Thanks,
> Claudio
>
>
> Internet Email Confidentiality Footer
> -----------------------------------------------------------------------------------------------------
> La presente comunicazione, con le informazioni in essa contenute e  
> ogni documento o file allegato, e' rivolta unicamente alla/e persona/ 
> e cui e' indirizzata ed alle altre da questa autorizzata/e a  
> riceverla. Se non siete i destinatari/autorizzati siete avvisati che  
> qualsiasi azione, copia, comunicazione, divulgazione o simili basate  
> sul contenuto di tali informazioni e' vietata e potrebbe essere  
> contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia  
> di protezione dei dati personali). Se avete ricevuto questa  
> comunicazione per errore, vi preghiamo di darne immediata notizia al  
> mittente e di distruggere il messaggio originale e ogni file  
> allegato senza farne copia alcuna o riprodurne in alcun modo il  
> contenuto.
>
> This e-mail and its attachments are intended for the addressee(s)  
> only and are confidential and/or may contain legally privileged  
> information. If you have received this message by mistake or are not  
> one of the addressees above, you may take no action based on it, and  
> you may not copy or show it to anyone; please reply to this e-mail  
> and point out the error which has occurred.
> -----------------------------------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From odermann at googlemail.com  Fri Dec 19 08:44:49 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 19 Dec 2008 17:44:49 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
Message-ID: <5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>

it's me again about mod fax... it is short before christmas and my
whish is, to get mod fax working quite reliable. is this possible
under optimal conditions?

all our tests lead by far to more failed faxes than received faxes. i
really like the fax feature and would like to see it beeing usable.

is it just pure luck, if a fax was received or are there some
conditions out there, which could help beeing mod fax reliable?
second question: what about t38? will it come? is there chance, that
it will come? where are the difficulties with mod fax?

our fs servers are standing directly beside the sip switch of our
carrier. from the carriers switch, there is a 50 cm long cat6 cable
going into our cisco-switch. from the cisco switch there are 50 cm
long cat6 cables going into our fs servers.
i doubt, that there can be a signifant packet loss.
are there some settings, we could try out or is the faxing stuff just
unusable, till t38 is supported?


From anthony.minessale at gmail.com  Fri Dec 19 09:00:21 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Dec 2008 11:00:21 -0600
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
	<5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
Message-ID: <191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>

You don't know where the audio goes after that switch in the same room until
it gets to the guy
with the fax machine.

No it will not be improved by Christmas.  Not a chance.

Yes it will probably be much more reliable once it can do T38.

Be happy with what you have for the holiday season.



On Fri, Dec 19, 2008 at 10:44 AM, Dennis <odermann at googlemail.com> wrote:

> it's me again about mod fax... it is short before christmas and my
> whish is, to get mod fax working quite reliable. is this possible
> under optimal conditions?
>
> all our tests lead by far to more failed faxes than received faxes. i
> really like the fax feature and would like to see it beeing usable.
>
> is it just pure luck, if a fax was received or are there some
> conditions out there, which could help beeing mod fax reliable?
> second question: what about t38? will it come? is there chance, that
> it will come? where are the difficulties with mod fax?
>
> our fs servers are standing directly beside the sip switch of our
> carrier. from the carriers switch, there is a 50 cm long cat6 cable
> going into our cisco-switch. from the cisco switch there are 50 cm
> long cat6 cables going into our fs servers.
> i doubt, that there can be a signifant packet loss.
> are there some settings, we could try out or is the faxing stuff just
> unusable, till t38 is supported?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Fri Dec 19 09:09:32 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 19 Dec 2008 18:09:32 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
	<5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
	<191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
Message-ID: <5e414ed0812190909s458f791w65c77c273f2afb80@mail.gmail.com>

hi anthony,

thanks a lot for the clear answer. that is something i can work with :-)

i also want to thank you for the great support you gave us within the
last months and the great freeswitch. our fs servers are up and
running and everything works great (only fax is not working).

have a nice christmas (till i contact you because of some consulting
for final checks ;-)

dennis

2008/12/19 Anthony Minessale <anthony.minessale at gmail.com>:
> You don't know where the audio goes after that switch in the same room until
> it gets to the guy
> with the fax machine.
>
> No it will not be improved by Christmas.  Not a chance.
>
> Yes it will probably be much more reliable once it can do T38.
>
> Be happy with what you have for the holiday season.
>
>
>
> On Fri, Dec 19, 2008 at 10:44 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> it's me again about mod fax... it is short before christmas and my
>> whish is, to get mod fax working quite reliable. is this possible
>> under optimal conditions?
>>
>> all our tests lead by far to more failed faxes than received faxes. i
>> really like the fax feature and would like to see it beeing usable.
>>
>> is it just pure luck, if a fax was received or are there some
>> conditions out there, which could help beeing mod fax reliable?
>> second question: what about t38? will it come? is there chance, that
>> it will come? where are the difficulties with mod fax?
>>
>> our fs servers are standing directly beside the sip switch of our
>> carrier. from the carriers switch, there is a 50 cm long cat6 cable
>> going into our cisco-switch. from the cisco switch there are 50 cm
>> long cat6 cables going into our fs servers.
>> i doubt, that there can be a signifant packet loss.
>> are there some settings, we could try out or is the faxing stuff just
>> unusable, till t38 is supported?
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From odermann at googlemail.com  Fri Dec 19 09:33:02 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 19 Dec 2008 18:33:02 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
	<5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
	<191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
Message-ID: <5e414ed0812190933j772292bdw32bbb7213c6b6591@mail.gmail.com>

ahh, just a second. it seems that i did not realize a small
missunderstanding in you answer.

i do not want to SEND a fax, i just want to RECEIVE a fax. so the fax
comes in at out carrier and the rest is sent over about 1m of cat6 to
our fs server.

is there a difference or does it not matter, if we want to receive or
send a fax?





2008/12/19 Anthony Minessale <anthony.minessale at gmail.com>:
> You don't know where the audio goes after that switch in the same room until
> it gets to the guy
> with the fax machine.
>
> No it will not be improved by Christmas.  Not a chance.
>
> Yes it will probably be much more reliable once it can do T38.
>
> Be happy with what you have for the holiday season.
>
>
>
> On Fri, Dec 19, 2008 at 10:44 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> it's me again about mod fax... it is short before christmas and my
>> whish is, to get mod fax working quite reliable. is this possible
>> under optimal conditions?
>>
>> all our tests lead by far to more failed faxes than received faxes. i
>> really like the fax feature and would like to see it beeing usable.
>>
>> is it just pure luck, if a fax was received or are there some
>> conditions out there, which could help beeing mod fax reliable?
>> second question: what about t38? will it come? is there chance, that
>> it will come? where are the difficulties with mod fax?
>>
>> our fs servers are standing directly beside the sip switch of our
>> carrier. from the carriers switch, there is a 50 cm long cat6 cable
>> going into our cisco-switch. from the cisco switch there are 50 cm
>> long cat6 cables going into our fs servers.
>> i doubt, that there can be a signifant packet loss.
>> are there some settings, we could try out or is the faxing stuff just
>> unusable, till t38 is supported?
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mszlazak at aol.com  Fri Dec 19 11:30:42 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Fri, 19 Dec 2008 14:30:42 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
Message-ID: <8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>


?With the firewall ON or OFF the problem still remains.

I've tried 3 different set-ups in a dial plan extension.

1. With? only <action application="set" data="proxy_media=true"/> before bridging.
??? ??? ??? 
2. With only? <action application="set" data="bypass_media=true"/> before bridging.

3. Neither of the above in the extension.

Only 2 with "proxy-media=true" gets the audio across endpoints.

Help :-)


? 




 


 

-----Original Message-----
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, 19 Dec 2008 7:49 am
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?









It gives me the impression there is something wrong with your firewall running on the box.



Mike






On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:


I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.
 I'm trying to pass audio "internally" on the same machine between endpoints and have be advised that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 
 However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.
 Any comments?? 
 



Listen to 350+ music, sports, & news radio stations=2
0? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 _______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From jason at jasonjgw.net  Fri Dec 19 15:13:00 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Dec 2008 10:13:00 +1100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
Message-ID: <20081219231300.GA4413@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 11:20:57AM -0600, Anthony Minessale wrote:
> It seems to be related to 20ms vs 30ms ptime.
> 
> What are the 2 devices and what rev of FS are you on?
> 
> There was more code added in the last few weeks to smooth out this
> occurrence.

This might not be the same issue, but as an additional data point, I can
reliably generate audio drop-outs under revision 10725 as follows - it sounds
somewhat like a jittery network connection, with occasional glitches. Network
problems are not involved, though, as the phone and the machine running
FreeSWITCH are both on my desk, connected via my ADSL router.

On a Snom 320 SIP phone, select G.722 as the first codec.

Call extension 3001 in the default context of the supplied FreeSWITCH
configuration, with nobody else calling into the conference, and listen to the
resulting audio.

Changing the packet size from 20ms to 30ms in the phone's configuration and
repeating the test gave me the same result.

When I recompile FreeSWITCH and re-install, I'll test this again and post a
follow-up if the problem persists.



From brian at freeswitch.org  Fri Dec 19 15:28:44 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Dec 2008 17:28:44 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <20081219231300.GA4413@jdc.jasonjgw.net>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
Message-ID: <11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>

Riddle me this... what firmware are you running?

/b

On Dec 19, 2008, at 5:13 PM, Jason White wrote:

> On a Snom 320 SIP phone, select G.722 as the first codec.
>
> Call extension 3001 in the default context of the supplied FreeSWITCH
> configuration, with nobody else calling into the conference, and  
> listen to the
> resulting audio.



From jason at jasonjgw.net  Fri Dec 19 15:36:59 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Dec 2008 10:36:59 +1100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
Message-ID: <20081219233659.GA5134@jdc.jasonjgw.net>

On Fri, Dec 19, 2008 at 05:28:44PM -0600, Brian West wrote:
> Riddle me this... what firmware are you running?
My apologies - 7.1.30.



From brian at freeswitch.org  Fri Dec 19 15:41:02 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Dec 2008 17:41:02 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <20081219233659.GA5134@jdc.jasonjgw.net>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
Message-ID: <AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>

Please update to 7.1.33 or higher, also I need a pcap of your  
situation email me a link where I can wget it if off list... I need  
the rtp and sip traffic.   You can do it from the phone or from  
FreeSWITCH.
/b

On Dec 19, 2008, at 5:36 PM, Jason White wrote:

> My apologies - 7.1.30.



From brian at freeswitch.org  Fri Dec 19 15:41:28 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Dec 2008 17:41:28 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <20081219233659.GA5134@jdc.jasonjgw.net>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
Message-ID: <2ABAE1B0-B8B1-453D-8B51-661F8E2F5D9B@freeswitch.org>

Btw I have a 300, 320, 360, m3 and an 820 on the way now.  (I don't  
see this problem you're describing at all)

/b

On Dec 19, 2008, at 5:36 PM, Jason White wrote:

> My apologies - 7.1.30.



From jason at jasonjgw.net  Fri Dec 19 15:51:25 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Dec 2008 10:51:25 +1100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <2ABAE1B0-B8B1-453D-8B51-661F8E2F5D9B@freeswitch.org>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
	<2ABAE1B0-B8B1-453D-8B51-661F8E2F5D9B@freeswitch.org>
Message-ID: <20081219235125.GA5269@jdc.jasonjgw.net>

I just tested again, and I myself am now having trouble reproducing it. It
happened on multiple occasions yesterday evening, though.

It is also easier to reproduce with an actual connection to a remote
end-point, but that obviously complicates the situation with potential network
issues. I haven't heard it under G.711 though.

It can't be system load, as the machine has been mostly idle during my
testing/experimentation.



From jason at jasonjgw.net  Fri Dec 19 23:42:35 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Dec 2008 18:42:35 +1100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
	<AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>
Message-ID: <20081220074235.GA4559@jdc.jasonjgw.net>

On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
> Please update to 7.1.33 or higher, also I need a pcap of your  
> situation email me a link where I can wget it if off list... I need  
> the rtp and sip traffic.   You can do it from the phone or from  
> FreeSWITCH.

I've just upgraded FreeSWITCH to 10889, and I'm definitely having audio
issues, but so far only reproduced via remote connections. I'll experiment
with the jitter buffer in case it's network related, and I'll run more local
tests as well.

Yesterday I was experiencing the problem locally, as reported. 

If it persists and if it appears not to be jitter in the network traffic, I'll
run tcpdump and make the output available.

Thanks again for the advice.



From mszlazak at aol.com  Sat Dec 20 00:17:50 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Sat, 20 Dec 2008 03:17:50 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com><B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
Message-ID: <8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>

With the firewall ON or OFF the problem still remains.



I've tried 3 different set-ups in a dial plan extension.



1. With? only <action application="set" data="proxy_media=true"/> before bridging.

??? ??? ??? 

2. With only? <action application="set" data="bypass_media=true"/> before bridging.



3. Neither of the above in the extension.



Only 2 with "bypass-media=true" gets the audio across endpoints.



Help :-)


 


 

-----Original Message-----
From: mszlazak at aol.com
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, 19 Dec 2008 11:30 am
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?











?With the firewall ON or OFF the problem still remains.



I've tried 3 different set-ups in a dial plan extension.



1. With? only <action application="set" data="proxy_media=true"/> before bridging.

??? ??? ??? 

2. With only? <action application="set" data="bypass_media=true"/> before bridging.



3. Neither of the above in the extension.



Only 2 with "proxy-media=true" gets the audio across endpoints.



Help :-)





? 









 





 



-----Original Message-----

From: Michael Jerris <mike at jerris.com>

To: freeswitch-users at lists.freeswitch.org

Sent: Fri, 19 Dec 2008 7:49 am

Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?












It gives me the
 impression there is something wrong with your firewall running on the box.







Mike













On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:



I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.

 I'm trying to pass audio "internally" on the same machine between endpoints and have be advis
ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 

 However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.

 Any comments?? 

 





Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 

 _______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.
org








=



 






_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





 





Listen to 350+ music, sports, & news radio stations ? 
including songs for the holidays ? FREE while you browse. Start Listening Now! 



 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From jaugenstine at gmail.com  Sat Dec 20 11:04:54 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Sat, 20 Dec 2008 11:04:54 -0800
Subject: [Freeswitch-users]  mod_java.so load issue
Message-ID: <207e7a5e0812201104l6280ba16g265486f750f10604@mail.gmail.com>

I am installing Freeswitch on Fedora.  I was building/installing the
mod_java.so module and I encountered the following load issue:

2008-12-20 10:34:58 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_java.so
**/usr/local/freeswitch/mod/mod_java.so: invalid ELF header**

Is this a build issue?  I am assuming maybe there is a g++ option that is
set incorrectly but my searches on Google and looking at gcc docs have not
provided a solution.

Jonathan
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From mike at jerris.com  Sat Dec 20 11:33:47 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Dec 2008 14:33:47 -0500
Subject: [Freeswitch-users] mod_java.so load issue
In-Reply-To: <207e7a5e0812201104l6280ba16g265486f750f10604@mail.gmail.com>
References: <207e7a5e0812201104l6280ba16g265486f750f10604@mail.gmail.com>
Message-ID: <F009F62E-F171-449D-9B70-4E1090B1DBF0@jerris.com>

I would suggest cleaning and rebuilding the module.  If that doesn't  
work could we arrange access to the box so I can take a look?

Mike

On Dec 20, 2008, at 2:04 PM, "jonathan augenstine" <jaugenstine at gmail.com 
 > wrote:

> I am installing Freeswitch on Fedora.  I was building/installing the  
> mod_java.so module and I encountered the following load issue:
>
> 2008-12-20 10:34:58 [CRIT] switch_loadable_module.c:839  
> switch_loadable_module_load_file() Error Loading module /usr/local/ 
> freeswitch/mod/mod_java.so
> **/usr/local/freeswitch/mod/mod_java.so: invalid ELF header**
>
> Is this a build issue?  I am assuming maybe there is a g++ option  
> that is set incorrectly but my searches on Google and looking at gcc  
> docs have not provided a solution.
>
> Jonathan
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Sat Dec 20 12:34:36 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Dec 2008 14:34:36 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <20081220074235.GA4559@jdc.jasonjgw.net>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
	<AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>
	<20081220074235.GA4559@jdc.jasonjgw.net>
Message-ID: <28CA023D-4A27-48D5-8261-2D7D3A74400E@freeswitch.org>

Do the pcap and show me... because I do not have this issue and I have  
the exact same phone.

/b

On Dec 20, 2008, at 1:42 AM, Jason White wrote:

> On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
>> Please update to 7.1.33 or higher, also I need a pcap of your
>> situation email me a link where I can wget it if off list... I need
>> the rtp and sip traffic.   You can do it from the phone or from
>> FreeSWITCH.
>
> I've just upgraded FreeSWITCH to 10889, and I'm definitely having  
> audio
> issues, but so far only reproduced via remote connections. I'll  
> experiment
> with the jitter buffer in case it's network related, and I'll run  
> more local
> tests as well.
>
> Yesterday I was experiencing the problem locally, as reported.
>
> If it persists and if it appears not to be jitter in the network  
> traffic, I'll
> run tcpdump and make the output available.
>
> Thanks again for the advice.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason at jasonjgw.net  Sat Dec 20 23:21:11 2008
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Dec 2008 18:21:11 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <20081218091149.GA11826@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
	<5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
	<20081218085323.GA11635@jdc.jasonjgw.net>
	<0B5DCBAA-5FA5-488A-8189-39878A5FCDA0@freeswitch.org>
	<20081218091149.GA11826@jdc.jasonjgw.net>
Message-ID: <20081221072111.GA12815@jdc.jasonjgw.net>

The solution was to edit /etc/asterisk/sip.conf and change
bindaddr = 0.0.0.0
to
bindaddr = x.x.x.x
where x.x.x.x is, naturally, replaced by the host's actual IPv4 address.

Following this change to the Asterisk configuration, Asterisk can bind to port
5060 under IPv4, and FreeSWITCH can bind to port 5060 under IPv6.




From saigop at gmail.com  Sun Dec 21 07:09:41 2008
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Sun, 21 Dec 2008 20:39:41 +0530
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <DB682CA7-D304-4D78-ABA7-4BD91FF6A8AA@jerris.com>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
	<F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
	<2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>
	<DB682CA7-D304-4D78-ABA7-4BD91FF6A8AA@jerris.com>
Message-ID: <2ea4d47e0812210709q59ced653t15bde61cb9b4683e@mail.gmail.com>

Hi Micheal,
     Is it anything like i am violating the laws? please let me know.

On Fri, Dec 5, 2008 at 8:11 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Dec 5, 2008, at 6:23 AM, Gopalakrishnan A.N wrote:
>
> > Hi Micheal,
> >
> >   Thanks for the reply! cant I try with tone detect?
> >
> >   Like dial a number in session and try to detect with tone detect
> > and then bridge the call with some extension.
>
> If you know the exact frequency of the tone you can, but I suspect you
> do not.
>
> Mike
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thank you  with regards,
Gopal,
PeopleTech Systems Private Limited
www.peopletech.co.in
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From saigop at gmail.com  Sun Dec 21 07:12:58 2008
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Sun, 21 Dec 2008 20:42:58 +0530
Subject: [Freeswitch-users] sessions not ending up
Message-ID: <2ea4d47e0812210712o10ac8c28kd1222702d6f34c88@mail.gmail.com>

Hi,
    I have configured the freeswitch, we are dialing through event socket,
if i dial a call per day say around 200 to 300 calls, at the end of the day
the sessions are not ending up in the freeswitch, i can able to see in the
console till all the calls were hanged up, I am using .NET crm.

    Any suggestions would help us.

-- 
Thank you  with regards,
Gopal,
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From msc at freeswitch.org  Sun Dec 21 12:30:59 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Sun, 21 Dec 2008 12:30:59 -0800
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <2ea4d47e0812210709q59ced653t15bde61cb9b4683e@mail.gmail.com>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
	<F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
	<2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>
	<DB682CA7-D304-4D78-ABA7-4BD91FF6A8AA@jerris.com>
	<2ea4d47e0812210709q59ced653t15bde61cb9b4683e@mail.gmail.com>
Message-ID: <C71E023A-B1FE-43B6-98D5-A43850117B75@freeswitch.org>

I am not a lawyer so I can't tell you for sure. However, I'm not aware  
of any US laws against beep detection.

-MC

Sent from my iPhone

On Dec 21, 2008, at 7:09 AM, "Gopalakrishnan A.N" <saigop at gmail.com>  
wrote:

> Hi Micheal,
>
>      Is it anything like i am violating the laws? please let me know.
>
> On Fri, Dec 5, 2008 at 8:11 PM, Michael Jerris <mike at jerris.com>  
> wrote:
>
> On Dec 5, 2008, at 6:23 AM, Gopalakrishnan A.N wrote:
>
> > Hi Micheal,
> >
> >   Thanks for the reply! cant I try with tone detect?
> >
> >   Like dial a number in session and try to detect with tone detect
> > and then bridge the call with some extension.
>
> If you know the exact frequency of the tone you can, but I suspect you
> do not.
>
> Mike
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Thank you  with regards,
> Gopal,
> PeopleTech Systems Private Limited
> www.peopletech.co.in
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Sun Dec 21 13:12:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 21 Dec 2008 15:12:51 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <28CA023D-4A27-48D5-8261-2D7D3A74400E@freeswitch.org>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
	<AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>
	<20081220074235.GA4559@jdc.jasonjgw.net>
	<28CA023D-4A27-48D5-8261-2D7D3A74400E@freeswitch.org>
Message-ID: <191c3a030812211312u24f30d82j874dc918b16c4ffe@mail.gmail.com>

is it only the conference?
what if you call an extension playing a file instead.

On Sat, Dec 20, 2008 at 2:34 PM, Brian West <brian at freeswitch.org> wrote:

> Do the pcap and show me... because I do not have this issue and I have
> the exact same phone.
>
> /b
>
> On Dec 20, 2008, at 1:42 AM, Jason White wrote:
>
> > On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
> >> Please update to 7.1.33 or higher, also I need a pcap of your
> >> situation email me a link where I can wget it if off list... I need
> >> the rtp and sip traffic.   You can do it from the phone or from
> >> FreeSWITCH.
> >
> > I've just upgraded FreeSWITCH to 10889, and I'm definitely having
> > audio
> > issues, but so far only reproduced via remote connections. I'll
> > experiment
> > with the jitter buffer in case it's network related, and I'll run
> > more local
> > tests as well.
> >
> > Yesterday I was experiencing the problem locally, as reported.
> >
> > If it persists and if it appears not to be jitter in the network
> > traffic, I'll
> > run tcpdump and make the output available.
> >
> > Thanks again for the advice.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Sun Dec 21 14:49:14 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 21 Dec 2008 16:49:14 -0600
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
Message-ID: <191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>

Try adding more ip to your box and give each thing it's own dedicated
virtual IP.
Doing everything on the same box can be tricky.


On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:

> With the firewall ON or OFF the problem still remains.
>
> I've tried 3 different set-ups in a dial plan extension.
>
> 1. With  only <action application="set" data="proxy_media=true"/> before
> bridging.
>
> 2. With only  <action application="set" data="bypass_media=true"/> before
> bridging.
>
> 3. Neither of the above in the extension.
>
> Only 2 with "bypass-media=true" gets the audio across endpoints.
>
> Help :-)
>
>
>  -----Original Message-----
> From: mszlazak at aol.com
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, 19 Dec 2008 11:30 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media
> work?
>
>   With the firewall ON or OFF the problem still remains.
>
> I've tried 3 different set-ups in a dial plan extension.
>
> 1. With  only <action application="set" data="proxy_media=true"/> before
> bridging.
>
> 2. With only  <action application="set" data="bypass_media=true"/> before
> bridging.
>
> 3. Neither of the above in the extension.
>
> Only 2 with "proxy-media=true" gets the audio across endpoints.
>
> Help :-)
>
>
>
>
>
> 0A
>
>
>  -----Original Message-----
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, 19 Dec 2008 7:49 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media
> work?
>
>  It gives me the impression there is something wrong with your firewall
> running on the box.
>  Mike
>
>   On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>
> I find it strange that I can have to endpoints get audio went using bypass
> media mode but the audio fails to go across endpoints if I use proxy media
> mode.
> I'm trying to pass audio "internally" on the same machine between endpoints
> and have be advis ed that a reason the audio may fail to be passed is
> because there is some RTP timing and IP address/port issues.
> However, FS has no problem "connecting" ports if i change the mode to
> bypass media. This gives me the impression that something is wrong with FS
> proxy media mode.
> Any comments?
>
> ------------------------------
> Listen to 350+ music, sports, & news radio stations ? including songs for
> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
>  _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch. org <http://www.freeswitch.org>
>
>
>  =
>
> _______________________________________________
>
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> ------------------------------
> Listen to 350+ music, sports, & news radio stations ? including songs for
> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> Listen to 350+ music, sports, & news radio stations ? including songs for
> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From can_man at gmx.de  Sun Dec 21 15:55:54 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 22 Dec 2008 00:55:54 +0100
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one does not
	register
Message-ID: <20081221235554.311330@gmx.net>

Hello,

I have two sipgate.de accounts and would like to attache them both at the same time to freeswitch. 
My problem is that I am not sure how to dial out using the second account. For the first one I can do: 

<action application="bridge" data="sofia/gateway/sipgate.de/$1 at sipgate.de"/>

and everything works, but for the second one I always get: [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g. "sofia/gateway/sipgate2/$1 at sipgate.de" or "sofia/gateway/sipgate.de/$1 at sipgate2"
 
My sip profile looks like this:

<include>
  <gateway name="sipgate.de">
    <param name="realm" value="sipgate.de"/>
    <param name="proxy" value="sipgate.de"/>
    <param name="username" value="******"/>
    <param name="password" value="******"/>
    <param name="extension" value="10000"/>
  </gateway>
</include>

<include>
  <gateway name="sipgate2">
    <param name="realm" value="sipgate.de"/>
    <param name="proxy" value="sipgate.de"/>
    <param name="username" value="******"/>
    <param name="password" value="******"/>
    <param name="extension" value="10000"/>
  </gateway>
</include>

I tried to change gateway name with realm, but no luck.


Thank you very much for your help.

Phil
-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From brian at freeswitch.org  Sun Dec 21 16:56:50 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Dec 2008 18:56:50 -0600
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one does not
	register
In-Reply-To: <20081221235554.311330@gmx.net>
References: <20081221235554.311330@gmx.net>
Message-ID: <3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>

Show me the full extension.

/b

On Dec 21, 2008, at 5:55 PM, can_man at gmx.de wrote:

> <action application="bridge" data="sofia/gateway/sipgate.de/$1 at sipgate.de 
> "/>
>
> and everything works, but for the second one I always get:  
> [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g. "sofia/gateway/sipgate2/$1 at sipgate.de 
> " or "sofia/gateway/sipgate.de/$1 at sipgate2"

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From daldworth at teliax.com  Sun Dec 21 17:37:11 2008
From: daldworth at teliax.com (David Aldworth)
Date: Sun, 21 Dec 2008 18:37:11 -0700
Subject: [Freeswitch-users] Setting codec/dtmf mode
Message-ID: <4F2EBDE9-9EB1-4696-9DFC-651F76BB0EC5@teliax.com>

I'm looking for the most effective way to make sure I'm always forcing  
inband dtmf and PCMU on the PSTN <-> FS side of inbound and outbound  
calls. FS is always in the middle of the media. The FS <-> SIP UA  
(customer) side will be rfc2833 and whatever the negotiated codec for  
that particular UA happens to be. I know I can set <param name="codec- 
prefs" value="PCMU"/> and <param name="inbound-codec-negotiation"  
value="greedy"/> in the internal sip profile but won't the external  
sip profile settings override this when UA dial out? (they hit the  
external profile first in this case)

I'm basically fishing for suggestions on the best way to use start/ 
stop_dtmf for the inband detection and start/stop_dtmf_generate for  
sending the dtmf.

In asterisk this would have been accomplished by setting up separate  
stanza's in sip.conf and setting the dtmfmode= and allow= line per the  
respective legs of the calls. So, calls coming to/from the PSTN would  
have dtmfmode=inband and allow=ulaw, meanwhile UA's connecting to  
asterisk would have dtmfmode=rfc2833 and allow=ulaw, gsm, etc.

Why on earth would I be doing this? Well, in the interest of keeping  
the explanation short, we are limited to the common denominator of all  
our upstream PSTN carriers and they (or their equipment rather) always  
support this setup.

Thanks for any advice.

David



From jason at jasonjgw.net  Sun Dec 21 21:58:16 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Dec 2008 16:58:16 +1100
Subject: [Freeswitch-users] FreeSWITCH port audio module
Message-ID: <20081222055816.GA14532@jdc.jasonjgw.net>

One of the valuable features of FreeSWITCH is that it can be used as a soft
phone, as described on the wiki. In testing this, I discovered the following
issues, any comments on which would be welcome.

1. FreeSWITCH>pa rescan
-ERR no reply

FreeSWITCH>

After plugging in a USB head set and running this command, the new device
wasn't enumerated by pa devlist. Restarting FreeSWITCH of course solved it,
however.

2. More seriously, when using port audio (with a head set as the audio device,
in case that's significant), I'm hearing digital distortion (clipped samples?)
when the other party speaks slightly more loudly than usual. 

When calling via a Snom phone (rather than PortAudio) I have only experienced
this distortion when using FreeSWITCH under G.722, but I'll have to do more
testing to identify the exact combinations that produce it and those which
don't. I haven't heard it with an 8khz call from a SIP phone, so it does
appear to be a FreeSWITCH issue to some extent. I haven't been able to
eliminate it by adjusting the Alsa settings of my audio device.

3. I've encountered errors while trying to access an Intel HDA sound card with
FreeSWITCH, whereby PortAudio fails to open the audio device. Setting the
sample rate in portaudio.conf.xml to 48 khz may have contributed to the
solution, but there have been other changes to my system as well (including a
FreeSWITCh upgrade to revision 10889.

Some sound cards only suport 48 khz, apparently, so if others have problems, I
would suggest adjusting the sample rate in the configuration as I did.



From mszlazak at aol.com  Sun Dec 21 22:44:57 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 22 Dec 2008 01:44:57 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com><B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com><8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com><8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
Message-ID: <8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>


 Hi Anthony,

I actually suggested adding IP's to a Voxeo-Prophecy support person before but they thought that could be problematic. I went along with the earlier warning but now you have suggested it again. What makes everything on the same box tricky? 

Also, the thing that surprises me a bit is that bypass-media works but proxy-media or the default doesn't. 

Would you be kind enough to elaborate.

Thanks. Mark.



 


 

-----Original Message-----
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Sun, 21 Dec 2008 2:49 pm
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?









Try adding more ip to your box and give each thing it's own dedicated virtual IP.
Doing everything on the same box can be tricky.



On Sat, Dec 20, 2008 at 2:17 AM,  <mszlazak at aol.com> wrote:



With the firewall ON or OFF the problem still remains.






I've tried 3 different set-ups in a dial plan extension.





1. With? only <action application="set" data="proxy_media=true"/> before bridging.


??? ??? ??? 


2. With only? <action application="set" data="bypass_media=true"/> before bridging.





3. Neither of the above in the extension.






Only 2 with "bypass-media=true" gets the audio across endpoints.





Help :-)





 





 



-----Original Message-----

From: mszlazak at aol.com

To: freeswitch-users at lists.freeswitch.org




Sent: Fri, 19 Dec 2008 11:30 am

Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?

















?With the firewall ON or OFF the problem still remains.





I've tried 3 different set-ups in a dial plan extension.





1. With? only <action application="set" data="proxy_media=true"/> before bridging.


??? ??? ??? 


2. With only? <action application="set" data="bypass_media=true"/> before bridging.





3. Neither of the above in the extension.





Only 2 with "proxy-media=true" gets the audio across endpoints.





Help :-)








? 










0A







 








 





-----Original Message-----


From: Michael Jerris <mike at jerris.com>


To: freeswitch-users at lists.freeswitch.org


Sent: Fri, 19 Dec 2008 7:49 am


Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?















It gives me the impression there is something wrong with your firewall running on the box.











Mike




















On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:




I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.



 I'm trying to pass audio "internally" on the same machine between endpoints and have be a
dvis
ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 


 However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.


 Any comments?? 


 







Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 



 _______________________________________________



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Freeswitch-users at lists.freeswitch.org


http://lists.freeswitch.org/mailman/listinfo/freeswitch-users


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http://www.freeswitch.
org












=




 







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Freeswitch-users at lists.freeswitch.org



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/


AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

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From carole.olivier at enst.fr  Sun Dec 21 23:25:33 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Sun, 21 Dec 2008 23:25:33 -0800 (PST)
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <494A9EE2.7050507@freeswitch.org>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
	<21054080.post@talk.nabble.com> <49491847.3030403@lists.rupa.com>
	<21069519.post@talk.nabble.com> <494A9EE2.7050507@freeswitch.org>
Message-ID: <21123756.post@talk.nabble.com>


Hello,

Ray, thanks for your detailed answer!!

Fidibus83, when are you pressing *1 ? Because the last version of the
program I gave is thought to be executed like the following:
 - A and B are on call
 - A and B decide to change their call into a conference in order to invite
other people. So A or B press first *2. In this way, both legs are
transferred to the extension named transf_both_legs. In this extension, the
name of the conference will be created and added in the database so you can
not skip it.
 - A and B are now in a conference
 - A (or B) puts B (or A) on hold.
 - A (or B) calls C.
 - if C answers and agrees then A (or B) press *1. This transfers the
opposite leg, this means C, to the extension transf_opposite_leg. C will be
here transferred to the conference whose name will also be added in the db
as an entry for C.
 - A (or B) comes back to the conference it had put on hold.
 - A, B and C are now all in the conference.

So I am not sure but maybe you press first *1 instead of *2 isn't?
If this is not the case then you could press F8 to debug and find out what's
wrong. I am sorry but I can not help you more, I am new with freeswitch and
not very experienced.

Thanks to all of you for your help,
I am going to improve this small program,
Best regards,
Carole


Raymond Chandler-2 wrote:
> 
> Carole O. wrote:
>> Hello,
>>
>> Thanks for your answers!
>> Concerning the creation of a new variable for the conference the problem
>> is
>> that I do not create channels from the conference. I call separately a
>> new
>> member on a new channel and add it on the conference only if he agrees to
>> enter it. So it was the same problem as for the uuid, I am not sure I can
>> access the good variable from anywhere in case many conferences are
>> running.
>>
>>   
> you could use the db app to hold state across multiple calls... maybe 
> use the ${caller_id_number} and the ${destination_number} as keys for 
> the insert/select so that there's something constant to use in the 
> select... and another extension or two may be needed...
> 
> You could do the db lookup before you make the call so that you see if 
> your caller is already a member of a conference.... if he is, then the 
> transfer from *1 would work much the same as it does now except you'd 
> use the result of the db lookup as the conference number... if he's not 
> a member of an existing conference, then you could generate the uuid 
> like Anthony said before, then do a db insert for ${caller_id_number} 
> and ${destination_number} to insert that newly created uuid and use it 
> as the conference number.... one caveat that i see here is that the 
> destination_number would have to be exactly the same as if that user 
> were callling and it was his caller_id_number, otherwise your query will 
> fail.
> 
> you'll also need to "clean" the db when you hangup, which should be able 
> to be accomplished with an execute_on_hangup that does a delete of the 
> conf data for each user
> 
> -Ray
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21123756.html
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From Claudio.Cavalera at italtel.it  Mon Dec 22 00:28:01 2008
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Mon, 22 Dec 2008 09:28:01 +0100
Subject: [Freeswitch-users] Problem compiling socket2me
In-Reply-To: <94326D53-5AFD-4B71-845D-41F486D02D10@jerris.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0969643@BESONE.corp.dom>

freeswitch-users-bounces at lists.freeswitch.org wrote:
> mod_fax replaces socket2me, you don't need it anymore.
> 
> Mike

Ok thanks,
I would suggest to remove socket2me from trunk if still present.
Ciao,
Claudio


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From can_man at gmx.de  Mon Dec 22 03:06:02 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 22 Dec 2008 12:06:02 +0100
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one does
	not	register
In-Reply-To: <3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
Message-ID: <20081222110602.63410@gmx.net>



> Show me the full extension.

This extension with sipgate.de works for the single number specified, however when I change it to sipgate2 it doesn't. 

   <extension name="Long Distance">
    <condition field="destination_number" expression="^(0708351963)$">
     <action application="set" data="effective_caller_id_name=John Freeswitch"/>
     <action application="set" data="effective_caller_id_number=07083970139"/>
     <action application="bridge" data="sofia/gateway/sipgate.de/$1 at sipgate.de"/>
    </condition>
   </extension>

Thanks for your help.

Phil

> 
> On Dec 21, 2008, at 5:55 PM, can_man at gmx.de wrote:
> 
> > <action application="bridge"
> data="sofia/gateway/sipgate.de/$1 at sipgate.de 
> > "/>
> >
> > and everything works, but for the second one I always get:  
> > [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g.
> "sofia/gateway/sipgate2/$1 at sipgate.de 
> > " or "sofia/gateway/sipgate.de/$1 at sipgate2"
> 

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From pieter_eduard at biznetnetworks.com  Mon Dec 22 05:19:38 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Mon, 22 Dec 2008 20:19:38 +0700
Subject: [Freeswitch-users] call failed from PLMN to enum number
Message-ID: <494F93EA.4080608@biznetnetworks.com>

Hi,

I have an enum number, if I call the number from any ip extension ( i 
use default enum.conf that points to e164.arpa)  then the call goes well 
to my ATA that registers to my fs box,
but if i try to call the number from PLMN, i get the ring at my ATA and 
if i pick it up, there's no sound.

here's my public.xml config :

 <extension name="enum">
        <condition field="destination_number" expression="^12345678">
        <action application="set" data="proxy_media=true"/>
        <action application="bridge" data="sofia/default/1000%1.2.3.4"/>
        </condition>
    </extension>

For more detailed debug log, i already submit it on jira : 
http://jira.freeswitch.org/browse/MODAPP-186


regards,

-Pieter-



From carole.olivier at enst.fr  Mon Dec 22 05:35:44 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Mon, 22 Dec 2008 05:35:44 -0800 (PST)
Subject: [Freeswitch-users]  close channels properly
Message-ID: <21127913.post@talk.nabble.com>


Hello,

I use the following code to call VoIP speakers and make an announcement:
<extension name="security_announcement">
       <condition field="destination_number" expression="^9001$">
       <action application="set"
data="conference_auto_outcall_caller_id_name=security_announcement"/>
       <action application="set"
data="conference_auto_outcall_id_number=9001"/>
       <action application="set" data="conference_auto_outcall_timeout=60"/>
       <action application="set" data="conference_auto_outcall_flags=mute"/>
       <action application="set"
data="conference_auto_outcall_announce=announcement/announcement_security_delay.wav"/>
       <action application="conference_set_auto_outcall"
data="user/1021@$${domain}"/>
       <action application="conference_set_auto_outcall"
data="user/1022@$${domain}"/>
       <action application="conference"
data="conf_speaker at profilespeaker_announcement"/>
       </condition>
     </extension>

1021 and 1022 are the speakers.
At the end of the announcement, since there is no noise anymore, the
speakers stop listening but they do not send any messages to tell Freeswitch
it can close the opened channels. The channels are closed only after a
timeout of 5 minutes.

Does anybody know how I could force freeswitch to close all the channels
after the announcement? I have seen there is the application sched_hangup
but when I used it it only closes the channel to the caller and not the
other ones.

Thanks a lot for your help and Happy Holidays!
Carole
-- 
View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21127913.html
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From can_man at gmx.de  Mon Dec 22 06:05:32 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 22 Dec 2008 15:05:32 +0100
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one
	does	not	register RESOLVED
In-Reply-To: <20081222110602.63410@gmx.net>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
	<20081222110602.63410@gmx.net>
Message-ID: <20081222140532.264490@gmx.net>

Hello,

I know, no top posting, but I want to say that I resolved my problem.
Don't ask me why exactly it works now, but after I removed the effective caller_id_number it works like  this:


   <extension name="Long Distance">
    <condition field="destination_number" expression="^(0708351963)$">
    <action application="bridge" data="sofia/gateway/sipgate2/$1 at sipgate.de"/>
    </condition>
   </extension>


<include>
  <gateway name="sipgate.de">
    <param name="realm" value="sipgate.de"/>
    <param name="proxy" value="sipgate.de"/>
    <param name="username" value="******"/>
    <param name="password" value="******"/>
    <param name="extension" value="10000"/>
    <param name="register" value="true"/>
  </gateway>
</include>

<include>
  <gateway name="sipgate2">
    <param name="realm" value="sipgate.de"/>
    <param name="proxy" value="sipgate.de"/>
    <param name="username" value="******"/>
    <param name="password" value="******"/>
    <param name="extension" value="10000"/>
    <param name="register" value="true"/>
  </gateway>
</include>


I have added a note to the wiki Tested_Phone_Providers_Listing under Sipgate.de


Phil


> 
> 
> > Show me the full extension.
> 
> This extension with sipgate.de works for the single number specified,
> however when I change it to sipgate2 it doesn't. 
> 
>    <extension name="Long Distance">
>     <condition field="destination_number" expression="^(0708351963)$">
>      <action application="set" data="effective_caller_id_name=John
> Freeswitch"/>
>      <action application="set"
> data="effective_caller_id_number=07083970139"/>
>      <action application="bridge"
> data="sofia/gateway/sipgate.de/$1 at sipgate.de"/>
>     </condition>
>    </extension>
> 
> Thanks for your help.
> 
> Phil
> 
> > 
> > On Dec 21, 2008, at 5:55 PM, can_man at gmx.de wrote:
> > 
> > > <action application="bridge"
> > data="sofia/gateway/sipgate.de/$1 at sipgate.de 
> > > "/>
> > >
> > > and everything works, but for the second one I always get:  
> > > [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g.
> > "sofia/gateway/sipgate2/$1 at sipgate.de 
> > > " or "sofia/gateway/sipgate.de/$1 at sipgate2"
> > 
> 
> -- 
> Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen:
> http://www.gmx.net/de/go/multimessenger
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From brian at freeswitch.org  Mon Dec 22 06:35:23 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:35:23 -0600
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <494F93EA.4080608@biznetnetworks.com>
References: <494F93EA.4080608@biznetnetworks.com>
Message-ID: <6FFEA244-18DB-4D39-AC63-AE2AEA1FAFCD@freeswitch.org>

First off I would try the latest SVN Trunk you're a bit behind.  
Secondly I would try without proxy media mode on.

/b

On Dec 22, 2008, at 7:19 AM, Pieter Eduard wrote:

> Hi,
>
> I have an enum number, if I call the number from any ip extension ( i
> use default enum.conf that points to e164.arpa)  then the call goes  
> well
> to my ATA that registers to my fs box,
> but if i try to call the number from PLMN, i get the ring at my ATA  
> and
> if i pick it up, there's no sound.
>
> here's my public.xml config :
>
> <extension name="enum">
>        <condition field="destination_number" expression="^12345678">
>        <action application="set" data="proxy_media=true"/>
>        <action application="bridge" data="sofia/default/ 
> 1000%1.2.3.4"/>
>        </condition>
>    </extension>
>
> For more detailed debug log, i already submit it on jira :
> http://jira.freeswitch.org/browse/MODAPP-186
>
>
> regards,
>
> -Pieter-
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Dec 22 06:36:22 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:36:22 -0600
Subject: [Freeswitch-users] Problem compiling socket2me
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0969643@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0969643@BESONE.corp.dom>
Message-ID: <1104ACEC-491C-4CE6-B8CF-F7A01EC94618@freeswitch.org>

Its there to serve as an example of using the socket interface with  
audio if anything it should be updated to the latest SpanDSP code.  
Patches welcome.

/b

On Dec 22, 2008, at 2:28 AM, Cavalera Claudio Luigi wrote:

> Ok thanks,
> I would suggest to remove socket2me from trunk if still present.
> Ciao,
> Claudio



From brian at freeswitch.org  Mon Dec 22 06:37:17 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:37:17 -0600
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one does
	not	register
In-Reply-To: <20081222110602.63410@gmx.net>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
	<20081222110602.63410@gmx.net>
Message-ID: <CEDD95AF-BA95-48F1-8C6E-54B4536C68DF@freeswitch.org>

When dialing via a gateway DO NOT add the @sipgate.de to it.  That  
will cause it to fail.

/b

On Dec 22, 2008, at 5:06 AM, can_man at gmx.de wrote:

>
>   <extension name="Long Distance">
>    <condition field="destination_number" expression="^(0708351963)$">
>     <action application="set" data="effective_caller_id_name=John  
> Freeswitch"/>
>     <action application="set"  
> data="effective_caller_id_number=07083970139"/>
>     <action application="bridge" data="sofia/gateway/sipgate.de/$1 at sipgate.de 
> "/>
>    </condition>
>   </extension>

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From brian at freeswitch.org  Mon Dec 22 06:38:36 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:38:36 -0600
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one
	does	not	register RESOLVED
In-Reply-To: <20081222140532.264490@gmx.net>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
	<20081222110602.63410@gmx.net> <20081222140532.264490@gmx.net>
Message-ID: <66CA1BC2-9099-4B3D-82C3-18F3B013B521@freeswitch.org>

Remove the @sipgate.de it won't work properly if you add that.. you  
have this setting in the gateway xml so don't add it to the bridge  
line too.

/b

On Dec 22, 2008, at 8:05 AM, can_man at gmx.de wrote:

>    <action application="bridge" data="sofia/gateway/sipgate2/$1 at sipgate.de 
> "/>

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From brian at freeswitch.org  Mon Dec 22 06:39:17 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:39:17 -0600
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <21127913.post@talk.nabble.com>
References: <21127913.post@talk.nabble.com>
Message-ID: <2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>

What do you mean they close (hangup) after the 5 minute timeout?

/b

On Dec 22, 2008, at 7:35 AM, Carole O. wrote:

> 1021 and 1022 are the speakers.
> At the end of the announcement, since there is no noise anymore, the
> speakers stop listening but they do not send any messages to tell  
> Freeswitch
> it can close the opened channels. The channels are closed only after a
> timeout of 5 minutes.
>
> Does anybody know how I could force freeswitch to close all the  
> channels
> after the announcement? I have seen there is the application  
> sched_hangup
> but when I used it it only closes the channel to the caller and not  
> the
> other ones.



From kristjan.ugrin at gmail.com  Mon Dec 22 06:42:08 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Mon, 22 Dec 2008 15:42:08 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
Message-ID: <op.umki4gzjatcbto@linux.site>

I modified mod_dingaling.c so I can intercept google talk chat messages  
via socket - nothing fancy.
Then I wrote a java app that connects to freeswitch socket and in case of  
a proper message (trigger) it sends a command to freeswitch, in my case:
api originate sofia/default/1001 at 10.99.8.221  
&bridge(dingaling/gmail.com/my_mail at gmail.com)

Dingaling is logged in as another user which I have added as buddy, chat  
messages go trough, however when a call is started
between SIP and Gtalk client, we cannot hear each other at all.
Using freeswitch revision: 10866

Here is the log:
http://pastebin.com/m1eba2cb8

What can be the problem? First I thought it is because running sip client  
+ gtalk and freeswitch on one host, but then I
moved SIP phone and Gtalk to 2 different workstations, using the third  
only for freeswitch. Also calls from "call" example program
 from google lib works fine with same setup - something must be problematic  
with freeswitch, however cannot see what.

Thank you!

-- 
kriko


From brian at freeswitch.org  Mon Dec 22 06:44:51 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:44:51 -0600
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <494F93EA.4080608@biznetnetworks.com>
References: <494F93EA.4080608@biznetnetworks.com>
Message-ID: <CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>

I also need you to do this call again with "console loglevel debug" on  
and post it attached to the jira and not inline on the comments please.

/b
On Dec 22, 2008, at 7:19 AM, Pieter Eduard wrote:

> Hi,
>
> I have an enum number, if I call the number from any ip extension ( i
> use default enum.conf that points to e164.arpa)  then the call goes  
> well
> to my ATA that registers to my fs box,
> but if i try to call the number from PLMN, i get the ring at my ATA  
> and
> if i pick it up, there's no sound.
>
> here's my public.xml config :
>
> <extension name="enum">
>        <condition field="destination_number" expression="^12345678">
>        <action application="set" data="proxy_media=true"/>
>        <action application="bridge" data="sofia/default/ 
> 1000%1.2.3.4"/>
>        </condition>
>    </extension>
>
> For more detailed debug log, i already submit it on jira :
> http://jira.freeswitch.org/browse/MODAPP-186
>
>
> regards,
>
> -Pieter-
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From can_man at gmx.de  Mon Dec 22 06:52:56 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 22 Dec 2008 15:52:56 +0100
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one
	does	not	register
In-Reply-To: <CEDD95AF-BA95-48F1-8C6E-54B4536C68DF@freeswitch.org>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
	<20081222110602.63410@gmx.net>
	<CEDD95AF-BA95-48F1-8C6E-54B4536C68DF@freeswitch.org>
Message-ID: <20081222145256.184760@gmx.net>



> When dialing via a gateway DO NOT add the @sipgate.de to it.  That  
> will cause it to fail.
> 

Thank you. I have also updated the wiki.

Phil



-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From anthony.minessale at gmail.com  Mon Dec 22 07:19:08 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 09:19:08 -0600
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.umki4gzjatcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
Message-ID: <191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>

Your log shows rtp streams being allocated.
did you look at at the packets on the wire with a packet capture program?

You are better off using proper jingle and component mode.  What you are
describing sounds like
a workaround to avoid doing it right.



On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> I modified mod_dingaling.c so I can intercept google talk chat messages
> via socket - nothing fancy.
> Then I wrote a java app that connects to freeswitch socket and in case of
> a proper message (trigger) it sends a command to freeswitch, in my case:
> api originate sofia/default/1001 at 10.99.8.221
> &bridge(dingaling/gmail.com/my_mail at gmail.com)
>
> Dingaling is logged in as another user which I have added as buddy, chat
> messages go trough, however when a call is started
> between SIP and Gtalk client, we cannot hear each other at all.
> Using freeswitch revision: 10866
>
> Here is the log:
> http://pastebin.com/m1eba2cb8
>
> What can be the problem? First I thought it is because running sip client
> + gtalk and freeswitch on one host, but then I
> moved SIP phone and Gtalk to 2 different workstations, using the third
> only for freeswitch. Also calls from "call" example program
>  from google lib works fine with same setup - something must be problematic
> with freeswitch, however cannot see what.
>
> Thank you!
>
> --
> kriko
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec 22 07:24:53 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 09:24:53 -0600
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
	<8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
Message-ID: <191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>

I don't really know what your problem is.  I just saw you ask 3 times for
help and tried to offer a suggestion.
if you start FS with TPORT_LOG=1 you can see all the sip messages in the
console and you could
also run wireshark to look at a packet capture.

If you use the same IP for media on the same box for 3 programs at once you
may end up with 2 applictions choosing the same media port etc.

It's just a good practice to run every voip program on it's own IP.




On Mon, Dec 22, 2008 at 12:44 AM, <mszlazak at aol.com> wrote:

>  Hi Anthony,
>
> I actually suggested adding IP's to a Voxeo-Prophecy support person before
> but they thought that could be problematic. I went along with the earlier
> warning but now you have suggested it again. What makes everything on the
> same box tricky?
>
> Also, the thing that surprises me a bit is that bypass-media works but
> proxy-media or the default doesn't.
>
> Would you be kind enough to elaborate.
>
> Thanks. Mark.
>
>
>
>  -----Original Message-----
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sun, 21 Dec 2008 2:49 pm
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media
> work?
>
>  Try adding more ip to your box and give each thing it's own dedicated
> virtual IP.
> Doing everything on the same box can be tricky.
>
>
> On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:
>
>> With the firewall ON or OFF the problem still remains.
>>
>> I've tried 3 different set-ups in a dial plan extension.
>>
>> 1. With  only <action application="set" data="proxy_media=true"/> before
>> bridging.
>>
>> 2. With only  <action application="set" data="bypass_media=true"/> before
>> bridging.
>>
>> 3. Neither of the above in the extension.
>>
>>  Only 2 with "bypass-media=true" gets the audio across endpoints.
>>
>> Help :-)
>>
>>
>>  -----Original Message-----
>> From: mszlazak at aol.com
>> To: freeswitch-users at lists.freeswitch.org
>>  Sent: Fri, 19 Dec 2008 11:30 am
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>> Media work?
>>
>>    With the firewall ON or OFF the problem still remains.
>>
>> I've tried 3 different set-ups in a dial plan extension.
>>
>> 1. With  only <action application="set" data="proxy_media=true"/> before
>> bridging.
>>
>> 2. With only  <action application="set" data="bypass_media=true"/> before
>> bridging.
>>
>> 3. Neither of the above in the extension.
>>
>> Only 2 with "proxy-media=true" gets the audio across endpoints.
>>
>> Help :-)
>>
>>
>>
>>
>>
>>  0A
>>
>>
>>  -----Original Message-----
>> From: Michael Jerris <mike at jerris.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Fri, 19 Dec 2008 7:49 am
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>> Media work?
>>
>>  It gives me the impression there is something wrong with your firewall
>> running on the box.
>>  Mike
>>
>>   On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>>
>> I find it strange that I can have to endpoints get audio went using bypass
>> media mode but the audio fails to go across endpoints if I use proxy media
>> mode.
>> I'm trying to pass audio "internally" on the same machine between
>> endpoints and have be advis ed that a reason the audio may fail to be passed
>> is because there is some RTP timing and IP address/port issues.
>> However, FS has no problem "connecting" ports if i change the mode to
>> bypass media. This gives me the impression that something is wrong with FS
>> proxy media mode.
>> Any comments?
>>
>> ------------------------------
>> Listen to 350+ music, sports, & news radio stations ? including songs for
>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>
>>  _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch. org <http://www.freeswitch.org>
>>
>>
>>  =
>>
>> _______________________________________________
>>
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> ------------------------------
>> Listen to 350+ music, sports, & news radio stations ? including songs for
>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>
>>
>> _______________________________________________
>>
>>
>>
>> Freeswitch-users mailing list
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> Listen to 350+ music, sports, & news radio stations ? including songs for
>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<http://lists.freeswitch.org/mailman/listinfo++/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
>
> Freeswitch-users mailing list
> Freeswitch-users at lists
> .freeswitch.org <Freeswitch-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> Listen to 350+ music, sports, & news radio stations ? including songs for
> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ser at man.szczecin.pl  Mon Dec 22 05:54:40 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Mon, 22 Dec 2008 14:54:40 +0100
Subject: [Freeswitch-users] group call with BLF and pickup
Message-ID: <1229954080.9989.41.camel@worek.man.szczecin.pl>

Hi,

I'm currently evaluating FreeSWITCH to see if I can migrate to it from
Asterisk (which gives me now all functions I need, but has some problems
with predictability). I need only one atypical functionality:
1. we have one main extension, which clients call. lets name it 5555
2. when someone calls 5555, then immediately rings only one
   phone (1001) and 15 seconds later 3 more phones (1002-1004). phone
   1001 should be dialled once per call to 5555
3. all phones (1000-1010) have BLF monitoring 5555, so anyone can 
   pick it up when someone calls it

ad 3. it is acceptable that all phones monitor 1001 instead of 5555 to
pick up connection to 5555

After reading and googling about possible implementation in FS for few
hours I couldn't find anything useful. 

Could you tell if it is possible to implement in FS without artificial
limbs (i dunno if it's best English word for what I mean :) like I had
to do in Asterisk?

greetings,
Seweryn




From anthony.minessale at gmail.com  Mon Dec 22 07:39:57 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 09:39:57 -0600
Subject: [Freeswitch-users] FreeSWITCH port audio module
In-Reply-To: <20081222055816.GA14532@jdc.jasonjgw.net>
References: <20081222055816.GA14532@jdc.jasonjgw.net>
Message-ID: <191c3a030812220739y361b69e1kd45b9258733aff23@mail.gmail.com>

You did not clarify any of the details of your machine HW, OS etc.
The issue you reported with snoms are not reproducible based on your
description so you should
try harded to pinpoint it.  Telling us you have a problem but you are not
sure why etc is not helpful because now I have to stop and wait for you to
reply if I want to help you further.
Portaudio seems to not work very well under linux and we don't have any
linux machines with sound to test it with.

Can you please provide all the details including possibly access to your
machine so we can
reproduce your issue with your machine if we can't with our own.

Consider joining irc and informing us in real time

irc.freenode.net #freeswitch



On Sun, Dec 21, 2008 at 11:58 PM, Jason White <jason at jasonjgw.net> wrote:

> One of the valuable features of FreeSWITCH is that it can be used as a soft
> phone, as described on the wiki. In testing this, I discovered the
> following
> issues, any comments on which would be welcome.
>
> 1. FreeSWITCH>pa rescan
> -ERR no reply
>
> FreeSWITCH>
>
> After plugging in a USB head set and running this command, the new device
> wasn't enumerated by pa devlist. Restarting FreeSWITCH of course solved it,
> however.
>
> 2. More seriously, when using port audio (with a head set as the audio
> device,
> in case that's significant), I'm hearing digital distortion (clipped
> samples?)
> when the other party speaks slightly more loudly than usual.
>
> When calling via a Snom phone (rather than PortAudio) I have only
> experienced
> this distortion when using FreeSWITCH under G.722, but I'll have to do more
> testing to identify the exact combinations that produce it and those which
> don't. I haven't heard it with an 8khz call from a SIP phone, so it does
> appear to be a FreeSWITCH issue to some extent. I haven't been able to
> eliminate it by adjusting the Alsa settings of my audio device.
>
> 3. I've encountered errors while trying to access an Intel HDA sound card
> with
> FreeSWITCH, whereby PortAudio fails to open the audio device. Setting the
> sample rate in portaudio.conf.xml to 48 khz may have contributed to the
> solution, but there have been other changes to my system as well (including
> a
> FreeSWITCh upgrade to revision 10889.
>
> Some sound cards only suport 48 khz, apparently, so if others have
> problems, I
> would suggest adjusting the sample rate in the configuration as I did.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristjan.ugrin at gmail.com  Mon Dec 22 07:42:17 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Mon, 22 Dec 2008 16:42:17 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
Message-ID: <op.umklwrrxatcbto@linux.site>

There are absolutely no UDP packets going trough like when doing a call  
 from gtalk to gtalk.

You mean this (component mode):
http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F
Is there more documentation that this?

All I would like to do is to initiate a call between SIP telephone and  
gtalk user who typed in the message.

Thank you!


On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale  
<anthony.minessale at gmail.com> wrote:

> Your log shows rtp streams being allocated.
> did you look at at the packets on the wire with a packet capture program?
>
> You are better off using proper jingle and component mode.  What you are
> describing sounds like
> a workaround to avoid doing it right.
>
>
>
> On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> I modified mod_dingaling.c so I can intercept google talk chat messages
>> via socket - nothing fancy.
>> Then I wrote a java app that connects to freeswitch socket and in case  
>> of
>> a proper message (trigger) it sends a command to freeswitch, in my case:
>> api originate sofia/default/1001 at 10.99.8.221
>> &bridge(dingaling/gmail.com/my_mail at gmail.com)
>>
>> Dingaling is logged in as another user which I have added as buddy, chat
>> messages go trough, however when a call is started
>> between SIP and Gtalk client, we cannot hear each other at all.
>> Using freeswitch revision: 10866
>>
>> Here is the log:
>> http://pastebin.com/m1eba2cb8
>>
>> What can be the problem? First I thought it is because running sip  
>> client
>> + gtalk and freeswitch on one host, but then I
>> moved SIP phone and Gtalk to 2 different workstations, using the third
>> only for freeswitch. Also calls from "call" example program
>>  from google lib works fine with same setup - something must be  
>> problematic
>> with freeswitch, however cannot see what.
>>
>> Thank you!
>>
>> --
>> kriko
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>



-- 
Porn - the reason you need a new hard drive.


From gilbertandrew at me.com  Mon Dec 22 07:49:14 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Mon, 22 Dec 2008 10:49:14 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
 work?
In-Reply-To: <191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
	<8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
	<191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
Message-ID: <8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>

Mark,

Sorry I haven't had much time to help with this either.

But Anthony is offering good advice here. You are either going to have  
to work out what is going on at SIP/SDP/RTP level through logs and  
wireshark, or opt for a separate ip space. Another option (besides  
virtual ips) is VMWare or VirtualBox, although VMWare is probably  
easier to setup and bridge naturally to your host.

Vm's are just so easy anymore and it definitely seems like you are  
going against the grain right now.

Also - realizing you got here because of the need for ASR. I do have  
the Lumenvox license, and I was able to compile the module out of SVN.  
I have not tested anything yet. If things go well I should have some  
time after the 25th for this. My goal would be to get pizza or  
something akin to work.

Andy


On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:

> I don't really know what your problem is.  I just saw you ask 3  
> times for help and tried to offer a suggestion.
> if you start FS with TPORT_LOG=1 you can see all the sip messages in  
> the console and you could
> also run wireshark to look at a packet capture.
>
> If you use the same IP for media on the same box for 3 programs at  
> once you may end up with 2 applictions choosing the same media port  
> etc.
>
> It's just a good practice to run every voip program on it's own IP.
>
>
>
>
> On Mon, Dec 22, 2008 at 12:44 AM, <mszlazak at aol.com> wrote:
> Hi Anthony,
>
> I actually suggested adding IP's to a Voxeo-Prophecy support person  
> before but they thought that could be problematic. I went along with  
> the earlier warning but now you have suggested it again. What makes  
> everything on the same box tricky?
>
> Also, the thing that surprises me a bit is that bypass-media works  
> but proxy-media or the default doesn't.
>
> Would you be kind enough to elaborate.
>
> Thanks. Mark.
>
>
>
> -----Original Message-----
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sun, 21 Dec 2008 2:49 pm
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
> Proxy Media work?
>
> Try adding more ip to your box and give each thing it's own  
> dedicated virtual IP.
> Doing everything on the same box can be tricky.
>
>
> On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:
> With the firewall ON or OFF the problem still remains.
>
> I've tried 3 different set-ups in a dial plan extension.
>
> 1. With  only <action application="set" data="proxy_media=true"/>  
> before bridging.
>
> 2. With only  <action application="set" data="bypass_media=true"/>  
> before bridging.
>
> 3. Neither of the above in the extension.
>
> Only 2 with "bypass-media=true" gets the audio across endpoints.
>
> Help :-)
>
>
> -----Original Message-----
> From: mszlazak at aol.com
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, 19 Dec 2008 11:30 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
> Proxy Media work?
>
>  With the firewall ON or OFF the problem still remains.
>
> I've tried 3 different set-ups in a dial plan extension.
>
> 1. With  only <action application="set" data="proxy_media=true"/>  
> before bridging.
>
> 2. With only  <action application="set" data="bypass_media=true"/>  
> before bridging.
>
> 3. Neither of the above in the extension.
>
> Only 2 with "proxy-media=true" gets the audio across endpoints.
>
> Help :-)
>
>
>
>
>
> 0A
>
>
> -----Original Message-----
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, 19 Dec 2008 7:49 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
> Proxy Media work?
>
> It gives me the impression there is something wrong with your  
> firewall running on the box.
>
> Mike
>
> On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>
>> I find it strange that I can have to endpoints get audio went using  
>> bypass media mode but the audio fails to go across endpoints if I  
>> use proxy media mode.
>> I'm trying to pass audio "internally" on the same machine between  
>> endpoints and have be advis ed that a reason the audio may fail to  
>> be passed is because there is some RTP timing and IP address/port  
>> issues.
>> However, FS has no problem "connecting" ports if i change the mode  
>> to bypass media. This gives me the impression that something is  
>> wrong with FS proxy media mode.
>> Any comments?
>>
>> Listen to 350+ music, sports, & news radio stations ? including  
>> songs for the holidays ? FREE while you browse. Start Listening  
>> Now!
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch. org
>
> =
> _______________________________________________
>
>
>
>
>
>
>
> Freeswitch-users mailing list
>
>
>
>
>
>
>
> Freeswitch-users at lists.freeswitch.org
>
>
>
>
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
>
>
>
>
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
>  _______________________________________________
>
>
>
>
>
>
> Freeswitch-users mailing list
>
>
>
> Freeswitch-users at lists.freeswitch.org
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
> http://www.freeswitch.org
>
>
>
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists
> .freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Dec 22 07:55:52 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 09:55:52 -0600
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
	<8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
	<191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
	<8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
Message-ID: <191c3a030812220755k6141aa45g219039fa20d3b66f@mail.gmail.com>

Andy,

ping us when you are ready. we have a lumenvox version of the pizza thing
already
That's the one we started with.


On Mon, Dec 22, 2008 at 9:49 AM, Andrew Gilbert <gilbertandrew at me.com>wrote:

> Mark,
> Sorry I haven't had much time to help with this either.
>
> But Anthony is offering good advice here. You are either going to have to
> work out what is going on at SIP/SDP/RTP level through logs and wireshark,
> or opt for a separate ip space. Another option (besides virtual ips) is
> VMWare or VirtualBox, although VMWare is probably easier to setup and bridge
> naturally to your host.
>
> Vm's are just so easy anymore and it definitely seems like you are going
> against the grain right now.
>
> Also - realizing you got here because of the need for ASR. I do have the
> Lumenvox license, and I was able to compile the module out of SVN. I have
> not tested anything yet. If things go well I should have some time after the
> 25th for this. My goal would be to get pizza or something akin to work.
>
> Andy
>
>
> On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:
>
> I don't really know what your problem is.  I just saw you ask 3 times for
> help and tried to offer a suggestion.
> if you start FS with TPORT_LOG=1 you can see all the sip messages in the
> console and you could
> also run wireshark to look at a packet capture.
>
> If you use the same IP for media on the same box for 3 programs at once you
> may end up with 2 applictions choosing the same media port etc.
>
> It's just a good practice to run every voip program on it's own IP.
>
>
>
>
> On Mon, Dec 22, 2008 at 12:44 AM, <mszlazak at aol.com> wrote:
>
>>  Hi Anthony,
>>
>> I actually suggested adding IP's to a Voxeo-Prophecy support person before
>> but they thought that could be problematic. I went along with the earlier
>> warning but now you have suggested it again. What makes everything on the
>> same box tricky?
>>
>> Also, the thing that surprises me a bit is that bypass-media works but
>> proxy-media or the default doesn't.
>>
>> Would you be kind enough to elaborate.
>>
>> Thanks. Mark.
>>
>>
>>
>>  -----Original Message-----
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Sun, 21 Dec 2008 2:49 pm
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>> Media work?
>>
>>  Try adding more ip to your box and give each thing it's own dedicated
>> virtual IP.
>> Doing everything on the same box can be tricky.
>>
>>
>> On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:
>>
>>> With the firewall ON or OFF the problem still remains.
>>>
>>> I've tried 3 different set-ups in a dial plan extension.
>>>
>>> 1. With  only <action application="set" data="proxy_media=true"/> before
>>> bridging.
>>>
>>> 2. With only  <action application="set" data="bypass_media=true"/> before
>>> bridging.
>>>
>>> 3. Neither of the above in the extension.
>>>
>>>  Only 2 with "bypass-media=true" gets the audio across endpoints.
>>>
>>> Help :-)
>>>
>>>
>>>  -----Original Message-----
>>> From: mszlazak at aol.com
>>> To: freeswitch-users at lists.freeswitch.org
>>>  Sent: Fri, 19 Dec 2008 11:30 am
>>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>>> Media work?
>>>
>>>    With the firewall ON or OFF the problem still remains.
>>>
>>> I've tried 3 different set-ups in a dial plan extension.
>>>
>>> 1. With  only <action application="set" data="proxy_media=true"/> before
>>> bridging.
>>>
>>> 2. With only  <action application="set" data="bypass_media=true"/> before
>>> bridging.
>>>
>>> 3. Neither of the above in the extension.
>>>
>>> Only 2 with "proxy-media=true" gets the audio across endpoints.
>>>
>>> Help :-)
>>>
>>>
>>>
>>>
>>>
>>>  0A
>>>
>>>
>>>  -----Original Message-----
>>> From: Michael Jerris <mike at jerris.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Fri, 19 Dec 2008 7:49 am
>>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>>> Media work?
>>>
>>>  It gives me the impression there is something wrong with your firewall
>>> running on the box.
>>>  Mike
>>>
>>>   On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>>>
>>> I find it strange that I can have to endpoints get audio went using
>>> bypass media mode but the audio fails to go across endpoints if I use proxy
>>> media mode.
>>> I'm trying to pass audio "internally" on the same machine between
>>> endpoints and have be advis ed that a reason the audio may fail to be passed
>>> is because there is some RTP timing and IP address/port issues.
>>> However, FS has no problem "connecting" ports if i change the mode to
>>> bypass media. This gives me the impression that something is wrong with FS
>>> proxy media mode.
>>> Any comments?
>>>
>>> ------------------------------
>>> Listen to 350+ music, sports, & news radio stations ? including songs for
>>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>>
>>>  _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch. org <http://www.freeswitch.org>
>>>
>>>
>>>  =
>>>
>>> _______________________________________________
>>>
>>>
>>>
>>>
>>> Freeswitch-users mailing list
>>>
>>>
>>>
>>> Freeswitch-users at lists.freeswitch.org
>>>
>>>
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>>
>>>
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>>
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> ------------------------------
>>> Listen to 350+ music, sports, & news radio stations ? including songs for
>>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>>
>>>
>>> _______________________________________________
>>>
>>>
>>>
>>>
>>> Freeswitch-users mailing list
>>>
>>> Freeswitch-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> ------------------------------
>>> Listen to 350+ music, sports, & news radio stations ? including songs for
>>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<http://lists.freeswitch.org/mailman/listinfo++/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>>
>> Freeswitch-users mailing list
>> Freeswitch-users at lists
>> .freeswitch.org <Freeswitch-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> Listen to 350+ music, sports, & news radio stations ? including songs for
>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec 22 08:02:02 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 10:02:02 -0600
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.umklwrrxatcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
Message-ID: <191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>

are you doing the trace on the FS box too?
it says it's established RTP and bridging.

NO audio is 9.8/10 times a firewall issue.

typing in a message is not the right way to call someone on jingle.
That's the point.  In component mode you add the sip ext to your buddy list
and call them the normal way.  This has nothing to do with your audio issue
though so it's
not a big deal.

On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> There are absolutely no UDP packets going trough like when doing a call
>  from gtalk to gtalk.
>
> You mean this (component mode):
>
> http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F
> Is there more documentation that this?
>
> All I would like to do is to initiate a call between SIP telephone and
> gtalk user who typed in the message.
>
> Thank you!
>
>
> On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>
> > Your log shows rtp streams being allocated.
> > did you look at at the packets on the wire with a packet capture program?
> >
> > You are better off using proper jingle and component mode.  What you are
> > describing sounds like
> > a workaround to avoid doing it right.
> >
> >
> >
> > On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:
> >
> >> I modified mod_dingaling.c so I can intercept google talk chat messages
> >> via socket - nothing fancy.
> >> Then I wrote a java app that connects to freeswitch socket and in case
> >> of
> >> a proper message (trigger) it sends a command to freeswitch, in my case:
> >> api originate sofia/default/1001 at 10.99.8.221
> >> &bridge(dingaling/gmail.com/my_mail at gmail.com)
> >>
> >> Dingaling is logged in as another user which I have added as buddy, chat
> >> messages go trough, however when a call is started
> >> between SIP and Gtalk client, we cannot hear each other at all.
> >> Using freeswitch revision: 10866
> >>
> >> Here is the log:
> >> http://pastebin.com/m1eba2cb8
> >>
> >> What can be the problem? First I thought it is because running sip
> >> client
> >> + gtalk and freeswitch on one host, but then I
> >> moved SIP phone and Gtalk to 2 different workstations, using the third
> >> only for freeswitch. Also calls from "call" example program
> >>  from google lib works fine with same setup - something must be
> >> problematic
> >> with freeswitch, however cannot see what.
> >>
> >> Thank you!
> >>
> >> --
> >> kriko
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
>
>
>
> --
> Porn - the reason you need a new hard drive.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristjan.ugrin at gmail.com  Mon Dec 22 08:19:50 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Mon, 22 Dec 2008 17:19:50 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
Message-ID: <op.umknnbetatcbto@linux.site>

But what I would like to achieve is something different (quite similar).
You type in a message like "call 1001 at 10.99.8.20" and you it would call a  
SIP buddy with any local number.

In component mode you need to add a buddy everytime for a different sip  
nr.?
Which would mean a lot of numbers if you would like to call more than one  
sip nr. in a lan.

As for the first issue, there are RTP packets traveling on FS, but never  
reach destination after they leave our internal network.
Do they have to go outside lan even when the call is placed in a lan  
between gtalk and SIP?
Gtalk to gtalk is no problem on same machines...


On Mon, 22 Dec 2008 17:02:02 +0100, Anthony Minessale  
<anthony.minessale at gmail.com> wrote:

> are you doing the trace on the FS box too?
> it says it's established RTP and bridging.
>
> NO audio is 9.8/10 times a firewall issue.
>
> typing in a message is not the right way to call someone on jingle.
> That's the point.  In component mode you add the sip ext to your buddy  
> list
> and call them the normal way.  This has nothing to do with your audio  
> issue
> though so it's
> not a big deal.
>
> On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> There are absolutely no UDP packets going trough like when doing a call
>>  from gtalk to gtalk.
>>
>> You mean this (component mode):
>>
>> http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F
>> Is there more documentation that this?
>>
>> All I would like to do is to initiate a call between SIP telephone and
>> gtalk user who typed in the message.
>>
>> Thank you!
>>
>>
>> On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>
>> > Your log shows rtp streams being allocated.
>> > did you look at at the packets on the wire with a packet capture  
>> program?
>> >
>> > You are better off using proper jingle and component mode.  What you  
>> are
>> > describing sounds like
>> > a workaround to avoid doing it right.
>> >
>> >
>> >
>> > On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com>  
>> wrote:
>> >
>> >> I modified mod_dingaling.c so I can intercept google talk chat  
>> messages
>> >> via socket - nothing fancy.
>> >> Then I wrote a java app that connects to freeswitch socket and in  
>> case
>> >> of
>> >> a proper message (trigger) it sends a command to freeswitch, in my  
>> case:
>> >> api originate sofia/default/1001 at 10.99.8.221
>> >> &bridge(dingaling/gmail.com/my_mail at gmail.com)
>> >>
>> >> Dingaling is logged in as another user which I have added as buddy,  
>> chat
>> >> messages go trough, however when a call is started
>> >> between SIP and Gtalk client, we cannot hear each other at all.
>> >> Using freeswitch revision: 10866
>> >>
>> >> Here is the log:
>> >> http://pastebin.com/m1eba2cb8
>> >>
>> >> What can be the problem? First I thought it is because running sip
>> >> client
>> >> + gtalk and freeswitch on one host, but then I
>> >> moved SIP phone and Gtalk to 2 different workstations, using the  
>> third
>> >> only for freeswitch. Also calls from "call" example program
>> >>  from google lib works fine with same setup - something must be
>> >> problematic
>> >> with freeswitch, however cannot see what.
>> >>
>> >> Thank you!
>> >>
>> >> --
>> >> kriko
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>>
>>
>>
>> --
>> Porn - the reason you need a new hard drive.
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>



-- 


From msc at freeswitch.org  Mon Dec 22 08:36:29 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 08:36:29 -0800
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <1229954080.9989.41.camel@worek.man.szczecin.pl>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>
Message-ID: <87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>

Seweryn,

Which phone(s) are you using? FS does BLF very well with Snom, Grandstream,
and Linksys. Also, the dialing scenario you mention is actually very easily
handled with FreeSWITCH. The devs are very clever and they set up a dialing
syntax mechanism that allows one to do all sorts of unique and even exotic
dialing setups.

Do you have a spare Linux machine and a few phones that you can do some
testing with? That would be the ideal way to get started quickly. If you do
have a Linux machine then the quickest way to get FS running is to do this:

http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install

Just remember one thing: FreeSWITCH is quite different from Asterisk, so
there is a bit of a learning curve, but it's totally worth it. :)

-MC

On Mon, Dec 22, 2008 at 5:54 AM, Seweryn Niemiec <ser at man.szczecin.pl>wrote:

> Hi,
>
> I'm currently evaluating FreeSWITCH to see if I can migrate to it from
> Asterisk (which gives me now all functions I need, but has some problems
> with predictability). I need only one atypical functionality:
> 1. we have one main extension, which clients call. lets name it 5555
> 2. when someone calls 5555, then immediately rings only one
>   phone (1001) and 15 seconds later 3 more phones (1002-1004). phone
>   1001 should be dialled once per call to 5555
> 3. all phones (1000-1010) have BLF monitoring 5555, so anyone can
>   pick it up when someone calls it
>
> ad 3. it is acceptable that all phones monitor 1001 instead of 5555 to
> pick up connection to 5555
>
> After reading and googling about possible implementation in FS for few
> hours I couldn't find anything useful.
>
> Could you tell if it is possible to implement in FS without artificial
> limbs (i dunno if it's best English word for what I mean :) like I had
> to do in Asterisk?
>
> greetings,
> Seweryn
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Dec 22 08:44:22 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 10:44:22 -0600
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>
	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
Message-ID: <BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>

The default config demo's this.. its using the db app to store the  
UUID and the intercept app to pickup the call.

/b

On Dec 22, 2008, at 10:36 AM, Michael Collins wrote:

> After reading and googling about possible implementation in FS for few
> hours I couldn't find anything useful.



From kawarod at laposte.net  Mon Dec 22 07:55:03 2008
From: kawarod at laposte.net (rod)
Date: Mon, 22 Dec 2008 19:55:03 +0400
Subject: [Freeswitch-users] SIP Headers and use of "" in the header
Message-ID: <494FB857.9090003@laposte.net>

Dear All,

I've been playing with the freeswitch options for one month now, and 
I've been able to use it with kamailio working as a registrar.

What I'd like to do is to add a diversion header using the following 
action in the dialplan:
    <action application="export" 
data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="unconditional""/>

Please note, that I'd like to put the word "unconditional" between 
quotes, this is to comply with the SIP gateway to which I'm sending trafic.

But I've been unable to set an escape character to use theses quotes, 
cause as you may understand, without escape character FS will consider 
this instruction instead:

<action application="export" 
data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="

cause the quote after reason= is considered as a closing quote for data=".

Is there a way to achieve this.

Thanks.
rod


From Laurent.Fabre at kirranet.com  Mon Dec 22 08:20:32 2008
From: Laurent.Fabre at kirranet.com (Laurent Fabre)
Date: Mon, 22 Dec 2008 17:20:32 +0100
Subject: [Freeswitch-users] Freeswitch/Sofia configuration problem
Message-ID: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>


Hello,

I've been trying to figure out for a few days why my freeswitch instance suddenly become insensitive to SIP packets without any warning.

What usually happen is the following :

1) start just fine in foreground mode and no errors
2) wait anywhere between 2 seconds and 20 minutes 
3) Sofia suddenly decide to reload everything for some reason
4) Sofia start processing SIP packets
5) work for an hour or so
6) Sofia suddenly decide to reload everything for some reason
7) become unresponsive again
8) goto 2

Both interfaces have public IP addresses assigned in a static manner (no DHCP).

I can see the SIP UDP & TCP requests comming from the phones on several sites on the wire.
The SIP TCP requests get RST in reply which is mean :(

There was a point in my setup where it would not happen but since I'm new to freeswitch I'm having an hard time backtracking.

I was wondering if iproute/tc and iptables were the culprits but I flushed everything (even rebooted without loading the rules) and it still doesn't work.

I thought some database was corrupt so I shutdown'd freeswitch and delete his db folder, no effect.

My server runs Debian 4.0etch for amd64, built freeswitch from SVN trunk.

Any pointers, help, cure against headaches would be great :)

Regards,

Laurent



From mike at jerris.com  Mon Dec 22 09:10:33 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Dec 2008 12:10:33 -0500
Subject: [Freeswitch-users] SIP Headers and use of "" in the header
In-Reply-To: <494FB857.9090003@laposte.net>
References: <494FB857.9090003@laposte.net>
Message-ID: <2BC8260A-5118-48E4-9F5B-60F5547A59D6@jerris.com>

You should be able to do this with xml CDATA syntax.

Mike

On Dec 22, 2008, at 10:55 AM, rod wrote:

> Dear All,
>
> I've been playing with the freeswitch options for one month now, and
> I've been able to use it with kamailio working as a registrar.
>
> What I'd like to do is to add a diversion header using the following
> action in the dialplan:
>    <action application="export"
> data
> ="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="unconditional""/>
>
> Please note, that I'd like to put the word "unconditional" between
> quotes, this is to comply with the SIP gateway to which I'm sending  
> trafic.
>
> But I've been unable to set an escape character to use theses quotes,
> cause as you may understand, without escape character FS will consider
> this instruction instead:
>
> <action application="export"
> data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="
>
> cause the quote after reason= is considered as a closing quote for  
> data=".



From brian at freeswitch.org  Mon Dec 22 09:11:26 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 11:11:26 -0600
Subject: [Freeswitch-users] SIP Headers and use of "" in the header
In-Reply-To: <494FB857.9090003@laposte.net>
References: <494FB857.9090003@laposte.net>
Message-ID: <DE122F9B-6648-4549-96FC-037BFF81BB98@freeswitch.org>

<action application="export"><![CDATA[sip_h_Diversion=<123456789 at 10.10.10.254 
 >;reason="unconditional"]]></action>

/b

On Dec 22, 2008, at 9:55 AM, rod wrote:

> Dear All,
>
> I've been playing with the freeswitch options for one month now, and
> I've been able to use it with kamailio working as a registrar.
>
> What I'd like to do is to add a diversion header using the following
> action in the dialplan:
>    <action application="export"
> data 
> ="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="unconditional""/>
>
> Please note, that I'd like to put the word "unconditional" between
> quotes, this is to comply with the SIP gateway to which I'm sending  
> trafic.
>
> But I've been unable to set an escape character to use theses quotes,
> cause as you may understand, without escape character FS will consider
> this instruction instead:
>
> <action application="export"
> data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="
>
> cause the quote after reason= is considered as a closing quote for  
> data=".
>
> Is there a way to achieve this.
>
> Thanks.
> rod
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Dec 22 09:30:46 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 11:30:46 -0600
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.umknnbetatcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
Message-ID: <191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>

if you see them leave FS and never reach dest.
It implies a firewall somewhere in between is blocking them.


On Mon, Dec 22, 2008 at 10:19 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> But what I would like to achieve is something different (quite similar).
> You type in a message like "call 1001 at 10.99.8.20" and you it would call a
> SIP buddy with any local number.
>
> In component mode you need to add a buddy everytime for a different sip
> nr.?
> Which would mean a lot of numbers if you would like to call more than one
> sip nr. in a lan.
>
> As for the first issue, there are RTP packets traveling on FS, but never
> reach destination after they leave our internal network.
> Do they have to go outside lan even when the call is placed in a lan
> between gtalk and SIP?
> Gtalk to gtalk is no problem on same machines...
>
>
> On Mon, 22 Dec 2008 17:02:02 +0100, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>
> > are you doing the trace on the FS box too?
> > it says it's established RTP and bridging.
> >
> > NO audio is 9.8/10 times a firewall issue.
> >
> > typing in a message is not the right way to call someone on jingle.
> > That's the point.  In component mode you add the sip ext to your buddy
> > list
> > and call them the normal way.  This has nothing to do with your audio
> > issue
> > though so it's
> > not a big deal.
> >
> > On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:
> >
> >> There are absolutely no UDP packets going trough like when doing a call
> >>  from gtalk to gtalk.
> >>
> >> You mean this (component mode):
> >>
> >>
> http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F
> >> Is there more documentation that this?
> >>
> >> All I would like to do is to initiate a call between SIP telephone and
> >> gtalk user who typed in the message.
> >>
> >> Thank you!
> >>
> >>
> >> On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>
> >> > Your log shows rtp streams being allocated.
> >> > did you look at at the packets on the wire with a packet capture
> >> program?
> >> >
> >> > You are better off using proper jingle and component mode.  What you
> >> are
> >> > describing sounds like
> >> > a workaround to avoid doing it right.
> >> >
> >> >
> >> >
> >> > On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com>
> >> wrote:
> >> >
> >> >> I modified mod_dingaling.c so I can intercept google talk chat
> >> messages
> >> >> via socket - nothing fancy.
> >> >> Then I wrote a java app that connects to freeswitch socket and in
> >> case
> >> >> of
> >> >> a proper message (trigger) it sends a command to freeswitch, in my
> >> case:
> >> >> api originate sofia/default/1001 at 10.99.8.221
> >> >> &bridge(dingaling/gmail.com/my_mail at gmail.com)
> >> >>
> >> >> Dingaling is logged in as another user which I have added as buddy,
> >> chat
> >> >> messages go trough, however when a call is started
> >> >> between SIP and Gtalk client, we cannot hear each other at all.
> >> >> Using freeswitch revision: 10866
> >> >>
> >> >> Here is the log:
> >> >> http://pastebin.com/m1eba2cb8
> >> >>
> >> >> What can be the problem? First I thought it is because running sip
> >> >> client
> >> >> + gtalk and freeswitch on one host, but then I
> >> >> moved SIP phone and Gtalk to 2 different workstations, using the
> >> third
> >> >> only for freeswitch. Also calls from "call" example program
> >> >>  from google lib works fine with same setup - something must be
> >> >> problematic
> >> >> with freeswitch, however cannot see what.
> >> >>
> >> >> Thank you!
> >> >>
> >> >> --
> >> >> kriko
> >> >>
> >> >> _______________________________________________
> >> >> Freeswitch-users mailing list
> >> >> Freeswitch-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Porn - the reason you need a new hard drive.
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
>
>
>
> --
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mszlazak at aol.com  Mon Dec 22 09:53:04 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 22 Dec 2008 12:53:04 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com><B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com><8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com><8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com><191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com><8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com><191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
	<8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
Message-ID: <8CB327545F492AF-928-221@WEBMAIL-MA02.sysops.aol.com>


 Hi Andy and Anthony.

Thanks Anthony for elaborating more and I'll attempt using another IP on the same box as well. 

Also, Prophecy support has asked me first to put one application on a separate box and then get some wireshark data so I'll attempt that also.

Andy, I didn't want to bother you given all those things you had to deal with. Welcome back.

I explored the VMware idea before but was warned that it would not work well with an ASR. This advice came from the Trixbox forums, LumenVox, FreeSwitch and Voxeo.

I understand that what I'm doing goes against the grain (i.e. voip) but frankly my target market really doesn't want anything to do with voip or even internet connectivity from their businesses. Plus there are other issues.

I'll let you know how it goes.

Happy holidays.

Mark.


 

-----Original Message-----
From: Andrew Gilbert <gilbertandrew at me.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 22 Dec 2008 7:49 am
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?









Mark,



Sorry I haven't had much time to help with this either.




But Anthony is offering good advice here.?You are either going to have to work out what is going on at SIP/SDP/RTP level through logs and wireshark, or opt for a separate ip space. Another option (besides virtual ips) is VMWare or VirtualBox, although VMWare is probably easier to setup and bridge naturally to your host.




Vm's are just so easy anymore and it definite
ly seems like you are going against the grain right now.




Also - realizing you got here because of the need for ASR. I do have the Lumenvox license, and I was able to compile the module out of SVN. I have not tested anything yet. If things go well I should have some time after the 25th for this. My goal would be to get pizza or something akin to work.




Andy







On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:


I don't really know what your problem is.? I just saw you ask 3 times for help and tried to offer a suggestion.
if you start FS with TPORT_LOG=1 you can see all the sip messages in the console and you could 
also run wireshark to look at a packet capture.
 
If you use the same IP for media on the same box for 3 programs at once you may end up with 2 applictions choosing the same media port etc.

It's just a good practice to run every voip program on it's own IP.
 




On Mon, Dec 22, 2008 at 12:44 AM,  <mszlazak at aol.com> wrote:

 
 Hi Anthony,
 
 I actually suggested adding IP's to a Voxeo-Prophecy support person before but they thought that could be problematic. I went along with the earlier warning but now you have suggested it again. What makes everything on the same box tricky? 
 
 Also, the thing that surprises me a bit is that bypass-media works but proxy-media or the default doesn't. 
 
 Would you be kind enough to elaborate.
 
 Thanks. Mark.
 
 

 
 
 
 
20
 
 -----Original Message-----
 From: Anthony Minessale <anthony.minessale at gmail.com>
 To: freeswitch-users at lists.freeswitch.org





 Sent: Sun, 21 Dec 2008 2:49 pm
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 Try adding more ip to your box and give each thing it's own dedicated virtual IP.
 Doing everything on the same box can be tricky.
 
 
 
On Sat, Dec 20, 2008 at 2:17 AM,  <mszlazak at aol.com> wrote:
 
 
With the firewall ON or OFF the problem still remains.
 
 I've tried 3 different set-ups in a dial plan extension.
 
 1. With? only <action application="set" data="proxy_media=true"/> before bridging.
 ??? ??? ??? 
 2. With only? <action application="set" data="bypass_media=true"/> before bridging.
 
 3. Neither of the above in the extension.
 
 
 Only 2 with "bypass-media=true" gets the audio across endpoints.
 
 Help :-)
  
 
 
 
 
 
 
 -----Original Message-----
 From: mszlazak at aol.com
 To: freeswitch-users at lists.freeswitch.org
 
 
 Sent: Fri, 19 Dec 2008 11:30 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 
 
 
?With the firewall ON or OFF the problem still remains.
 
 I've tried 3 different set-ups in a dial plan extension.
 
 1. With? only <action application="set" data="proxy_media=true"/> before bridging.
 ??? ??? ??? 
 2. With only? <action app
lication="set" data="bypass_media=true"/> before bridging.
 
 3. Neither of the above in the extension.
 
 Only 2 with "proxy-media=true" gets the audio across endpoints.
 
 Help :-)
 
 
 ? 
 
 
 
 
 0A 
 

 
 
 
 
 
 
 
 -----Original Message-----
 From: Michael Jerris <mike at jerris.com>
 To: freeswitch-users at lists.freeswitch.org
 Sent: Fri, 19 Dec 2008 7:49 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 It gives me the impression there is something wrong with your firewall running on the box. 

 
 
Mike
 

 
 
 
 
On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
 
 
I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.
  I'm trying to pass audio "internally" on the same machine between endpoints and have be advis ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 
  However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.
  Any comments?? 
   

 

Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
  _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lis
ts.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch. org
 

 
 
 = 
  
 
_______________________________________________







Freeswitch-users mailing list







Freeswitch-users at lists.freeswitch.org







http://lists.freeswitch.org/mailman/listinfo/freeswitch-users







UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users







http://www.freeswitch.org








 
  

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 
 

 
   
 
_______________________________________________






Freeswitch-users mailing list



Freeswitch-users at lists.freeswitch.org



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



http://www.freeswitch.org




 
   

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 

 
 

 -- 
 Anthony Minessale II
 
 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 
 AIM:
 anthm
 MSN:anthony_minessale at hotmail.com
 GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
 IRC: irc.freenode.net #freeswitch
 
 FreeSWITCH Developer Conference
 sip:888 at conference.freeswitch.org
 iax:guest at conference.freeswitch.org/888
 googletalk:conf+888 at conference.freeswitch.org
 pstn:213-799-1400
 
 
 
_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists
.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org


 
 



 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 


_______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
 
AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
 IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
 googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400
 _______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




=


 





_______
________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From anthony.minessale at gmail.com  Mon Dec 22 09:59:15 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 11:59:15 -0600
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812190933j772292bdw32bbb7213c6b6591@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
	<5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
	<191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
	<5e414ed0812190933j772292bdw32bbb7213c6b6591@mail.gmail.com>
Message-ID: <191c3a030812220959y1c4a754cu2c494e28f9324514@mail.gmail.com>

no difference.


On Fri, Dec 19, 2008 at 11:33 AM, Dennis <odermann at googlemail.com> wrote:

> ahh, just a second. it seems that i did not realize a small
> missunderstanding in you answer.
>
> i do not want to SEND a fax, i just want to RECEIVE a fax. so the fax
> comes in at out carrier and the rest is sent over about 1m of cat6 to
> our fs server.
>
> is there a difference or does it not matter, if we want to receive or
> send a fax?
>
>
>
>
>
> 2008/12/19 Anthony Minessale <anthony.minessale at gmail.com>:
> > You don't know where the audio goes after that switch in the same room
> until
> > it gets to the guy
> > with the fax machine.
> >
> > No it will not be improved by Christmas.  Not a chance.
> >
> > Yes it will probably be much more reliable once it can do T38.
> >
> > Be happy with what you have for the holiday season.
> >
> >
> >
> > On Fri, Dec 19, 2008 at 10:44 AM, Dennis <odermann at googlemail.com>
> wrote:
> >>
> >> it's me again about mod fax... it is short before christmas and my
> >> whish is, to get mod fax working quite reliable. is this possible
> >> under optimal conditions?
> >>
> >> all our tests lead by far to more failed faxes than received faxes. i
> >> really like the fax feature and would like to see it beeing usable.
> >>
> >> is it just pure luck, if a fax was received or are there some
> >> conditions out there, which could help beeing mod fax reliable?
> >> second question: what about t38? will it come? is there chance, that
> >> it will come? where are the difficulties with mod fax?
> >>
> >> our fs servers are standing directly beside the sip switch of our
> >> carrier. from the carriers switch, there is a 50 cm long cat6 cable
> >> going into our cisco-switch. from the cisco switch there are 50 cm
> >> long cat6 cables going into our fs servers.
> >> i doubt, that there can be a signifant packet loss.
> >> are there some settings, we could try out or is the faxing stuff just
> >> unusable, till t38 is supported?
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From wiltingtree at gmail.com  Mon Dec 22 09:34:19 2008
From: wiltingtree at gmail.com (Adam Wilt)
Date: Mon, 22 Dec 2008 12:34:19 -0500
Subject: [Freeswitch-users] Phone lines ring busy after awhile
Message-ID: <eb7797d70812220934v449243fhf6508bd892d92a0f@mail.gmail.com>

Hello.  I have an installation of FreeSwitch runnnig, and I wrote a Python
script to answer an inbound call and play an IVR.  It works fine for awhile,
but eventually it stops answering the phone line and the phone rings busy.
When I look in the FreeSwitch logs I don't see anything unusual. But I don't
think it's a problem with the VOIP phone service because when I restart
FreeSwitch the problem goes away.  Any ideas?
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From ser at man.szczecin.pl  Mon Dec 22 12:05:53 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Mon, 22 Dec 2008 21:05:53 +0100
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>
	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
Message-ID: <494FF321.1000207@man.szczecin.pl>

On Mon, Dec 22, 2008 at 5:36 PM, Michael Collins <msc at freeswitch.org> wrote:
> Which phone(s) are you using? FS does BLF very well with Snom, 
> Grandstream,

I have Grandstreams and I have already tested BLF and "standard"
pickup on FS. It works great on default configuration.

> and Linksys. Also, the dialing scenario you mention is actually very
> easily  handled with FreeSWITCH. The devs are very clever and they 
> set up a dialing syntax mechanism that allows one to do all sorts of
> unique and even exotic dialing setups.

are "bridge" and "sleep" actions + some FS's magic all what I need?

> Do you have a spare Linux machine and a few phones that you can do 
> some testing with? That would be the ideal way to get started 
> quickly. If you do
> have a Linux machine then the quickest way to get FS running is to do

First I tried to compile FS on my workstation (Ubuntu) but it has
failed due to too new libtool. But on latest Debian it was OK and I
have FS console under my fingers.

> Just remember one thing: FreeSWITCH is quite different from Asterisk,
>  so there is a bit of a learning curve, but it's totally worth it. :)

BTW: it would be nice to have some notes about reloading configs in
"getting started docs" on the wiki.

--
Best regards
Seweryn Niemiec


From ser at man.szczecin.pl  Mon Dec 22 12:14:26 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Mon, 22 Dec 2008 21:14:26 +0100
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
	<BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>
Message-ID: <494FF522.8030307@man.szczecin.pl>

On Mon, Dec 22, 2008 at 5:44 PM, Brian West <brian at freeswitch.org> wrote:
> The default config demo's this.. its using the db app to store the
 > UUID and the intercept app to pickup the call.

OK, first things first, can I monitor 5555 with BLF? Because if not,
then we get to the same point like in Asterisk, where I do:
- redirect 5555 to 1555 on upper tier (PSTN)
- "terminate" 5555 on the phone with 1001 extension
- group dial 5555 and 1002-1004 with delay when 1555 is called
Now 5555 can be monitored with BLF.

The second thing (important only when using above trick) is: when there
is a call to 1555 and phone 5555 is ringing, can I pick it up by **5555?

Because if not, then again we get to the same problem like in Asterisk. In
Asterisk when rings the phone with 5555 extension you have to pickup
1555 not 5555. So on BLF you monitor 5555 but to pickup you dial **1555.
This is quite stupid, users can't have two buttons to service one
extension (one for BLF and one for pickup). To get it working
transparently for endusers on Asterisk I had to hack and slash
1555 and **XXXX extensions (ugly global variables involved).

--
Best regards
Seweryn Niemiec


From msc at freeswitch.org  Mon Dec 22 12:16:08 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 12:16:08 -0800
Subject: [Freeswitch-users] Phone lines ring busy after awhile
In-Reply-To: <eb7797d70812220934v449243fhf6508bd892d92a0f@mail.gmail.com>
References: <eb7797d70812220934v449243fhf6508bd892d92a0f@mail.gmail.com>
Message-ID: <87f2f3b90812221216i42cbaeedx52cee1863e68667a@mail.gmail.com>

what operating system and which revision of FreeSWITCH?
-MC

On Mon, Dec 22, 2008 at 9:34 AM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hello.  I have an installation of FreeSwitch runnnig, and I wrote a Python
> script to answer an inbound call and play an IVR.  It works fine for awhile,
> but eventually it stops answering the phone line and the phone rings busy.
> When I look in the FreeSwitch logs I don't see anything unusual. But I don't
> think it's a problem with the VOIP phone service because when I restart
> FreeSwitch the problem goes away.  Any ideas?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Dec 22 12:18:23 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 14:18:23 -0600
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <494FF522.8030307@man.szczecin.pl>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
	<BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>
	<494FF522.8030307@man.szczecin.pl>
Message-ID: <6808A7E0-C0B2-4A8C-9D29-B045ABC37938@freeswitch.org>

The tip I can give you here is you have to set the  
"presence_id=5555 at domain" on a session so 1. the events are fired for  
that... and 2. you actually get the BLF for 5555  ;)  The presence ID  
will allow you to set arbitrary things you can then subscribe to on  
the phone.

/b

On Dec 22, 2008, at 2:14 PM, Seweryn Niemiec wrote:

> On Mon, Dec 22, 2008 at 5:44 PM, Brian West <brian at freeswitch.org>  
> wrote:
>> The default config demo's this.. its using the db app to store the
>> UUID and the intercept app to pickup the call.
>
> OK, first things first, can I monitor 5555 with BLF? Because if not,
> then we get to the same point like in Asterisk, where I do:
> - redirect 5555 to 1555 on upper tier (PSTN)
> - "terminate" 5555 on the phone with 1001 extension
> - group dial 5555 and 1002-1004 with delay when 1555 is called
> Now 5555 can be monitored with BLF.
>
> The second thing (important only when using above trick) is: when  
> there
> is a call to 1555 and phone 5555 is ringing, can I pick it up by  
> **5555?
>
> Because if not, then again we get to the same problem like in  
> Asterisk. In
> Asterisk when rings the phone with 5555 extension you have to pickup
> 1555 not 5555. So on BLF you monitor 5555 but to pickup you dial  
> **1555.
> This is quite stupid, users can't have two buttons to service one
> extension (one for BLF and one for pickup). To get it working
> transparently for endusers on Asterisk I had to hack and slash
> 1555 and **XXXX extensions (ugly global variables involved).
>
> --
> Best regards
> Seweryn Niemiec
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From wiltingtree at gmail.com  Mon Dec 22 12:27:08 2008
From: wiltingtree at gmail.com (Adam Wilt)
Date: Mon, 22 Dec 2008 15:27:08 -0500
Subject: [Freeswitch-users] Phone lines ring busy after awhile
Message-ID: <eb7797d70812221227o65eb07a4y141be8e5808989e1@mail.gmail.com>

 Sorry,  I'm using freeswitch-1.0.latest.tar.gz on Fedora.


>
> Message: 5
> Date: Mon, 22 Dec 2008 12:16:08 -0800
> From: "Michael Collins" <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] Phone lines ring busy after awhile
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>        <87f2f3b90812221216i42cbaeedx52cee1863e68667a at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> what operating system and which revision of FreeSWITCH?
> -MC
>
> On Mon, Dec 22, 2008 at 9:34 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
>
> > Hello.  I have an installation of FreeSwitch runnnig, and I wrote a
> Python
> > script to answer an inbound call and play an IVR.  It works fine for
> awhile,
> > but eventually it stops answering the phone line and the phone rings
> busy.
> > When I look in the FreeSwitch logs I don't see anything unusual. But I
> don't
> > think it's a problem with the VOIP phone service because when I restart
> > FreeSwitch the problem goes away.  Any ideas?
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
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From ser at man.szczecin.pl  Mon Dec 22 12:28:31 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Mon, 22 Dec 2008 21:28:31 +0100
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <6808A7E0-C0B2-4A8C-9D29-B045ABC37938@freeswitch.org>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>	<BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>	<494FF522.8030307@man.szczecin.pl>
	<6808A7E0-C0B2-4A8C-9D29-B045ABC37938@freeswitch.org>
Message-ID: <494FF86F.1030000@man.szczecin.pl>

Brian West wrote:
> The tip I can give you here is you have to set the  
> "presence_id=5555 at domain" on a session so 1. the events are fired for  
> that... and 2. you actually get the BLF for 5555  ;)  The presence ID  
> will allow you to set arbitrary things you can then subscribe to on  
> the phone.

that sounds cool. i'll start experiments tomorrow. thx for very fast reply.

-- 
greetings,
seweryn


From msc at freeswitch.org  Mon Dec 22 12:41:23 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 12:41:23 -0800
Subject: [Freeswitch-users] Phone lines ring busy after awhile
In-Reply-To: <eb7797d70812221227o65eb07a4y141be8e5808989e1@mail.gmail.com>
References: <eb7797d70812221227o65eb07a4y141be8e5808989e1@mail.gmail.com>
Message-ID: <87f2f3b90812221241w7dece8e2w54019ffeec0bead3@mail.gmail.com>

Hmmm... I think that might be a problem. I think that file hasn't been
getting updates - it looks way old on files.freeswitch.org.

Would you mind doing a few things to get your system on the latest? First,
mv /usr/local/freeswitch /usr/local/freeswitch.old. That will preserve your
existing config for future reference. Then do a new svn checkout and build
the latest, which is much improved. I recommend this process:

http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install

After you get the fresh install done then add back in your customizations
and try your script. Let us know how it goes.

-MC

On Mon, Dec 22, 2008 at 12:27 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Sorry,  I'm using freeswitch-1.0.latest.tar.gz on Fedora.
>
>
>>
>> Message: 5
>> Date: Mon, 22 Dec 2008 12:16:08 -0800
>> From: "Michael Collins" <msc at freeswitch.org>
>> Subject: Re: [Freeswitch-users] Phone lines ring busy after awhile
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID:
>>        <87f2f3b90812221216i42cbaeedx52cee1863e68667a at mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>>
>> what operating system and which revision of FreeSWITCH?
>> -MC
>>
>> On Mon, Dec 22, 2008 at 9:34 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>
>> > Hello.  I have an installation of FreeSwitch runnnig, and I wrote a
>> Python
>> > script to answer an inbound call and play an IVR.  It works fine for
>> awhile,
>> > but eventually it stops answering the phone line and the phone rings
>> busy.
>> > When I look in the FreeSwitch logs I don't see anything unusual. But I
>> don't
>> > think it's a problem with the VOIP phone service because when I restart
>> > FreeSwitch the problem goes away.  Any ideas?
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Mon Dec 22 12:48:05 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Dec 2008 15:48:05 -0500
Subject: [Freeswitch-users] Phone lines ring busy after awhile
In-Reply-To: <87f2f3b90812221241w7dece8e2w54019ffeec0bead3@mail.gmail.com>
References: <eb7797d70812221227o65eb07a4y141be8e5808989e1@mail.gmail.com>
	<87f2f3b90812221241w7dece8e2w54019ffeec0bead3@mail.gmail.com>
Message-ID: <A65C0A69-0771-46A8-B71F-0D1E65240031@jerris.com>

1.0.latest points to the last 1.0 release which was 1.0.1.

Mike

On Dec 22, 2008, at 3:41 PM, Michael Collins wrote:

> Hmmm... I think that might be a problem. I think that file hasn't  
> been getting updates - it looks way old on files.freeswitch.org.
>
> Would you mind doing a few things to get your system on the latest?  
> First, mv /usr/local/freeswitch /usr/local/freeswitch.old. That will  
> preserve your existing config for future reference. Then do a new  
> svn checkout and build the latest, which is much improved. I  
> recommend this process:
>
> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
>
> After you get the fresh install done then add back in your  
> customizations and try your script. Let us know how it goes.
>
> -MC
>
> On Mon, Dec 22, 2008 at 12:27 PM, Adam Wilt <wiltingtree at gmail.com>  
> wrote:
> Sorry,  I'm using freeswitch-1.0.latest.tar.gz on Fedora.
>
>
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From woof at nortel.com  Mon Dec 22 13:38:36 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 22 Dec 2008 16:38:36 -0500
Subject: [Freeswitch-users] Extra loud prompts when transcoded from L16@8000
	to G.722
Message-ID: <op.umk2emi1ssieo1@woof.us.nortel.com>

Woof!

I've noticed that the percieved volume of prompts recorded at L16 at 8000 is much louder (to the point of distortion) when played back via G.722 on Polycom phones, vs when played back via G.711.  The same prompts are also slightly louder when played back on SNOM phones via G.722 vs G.711, but not nearly as obnoxiously loud.  

I'm not sure if FS can do anything about this, it may be a Polycom issue, but is there a way to apply gain (negative in this case!) during the transcoding process so the percieved volume of the prompts is the same no matter which codec is selected?

--Woof!


From brian at freeswitch.org  Mon Dec 22 13:46:14 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 15:46:14 -0600
Subject: [Freeswitch-users] Extra loud prompts when transcoded from
	L16@8000 to G.722
In-Reply-To: <op.umk2emi1ssieo1@woof.us.nortel.com>
References: <op.umk2emi1ssieo1@woof.us.nortel.com>
Message-ID: <3A489B44-DFD2-4D1F-899E-9E7CFEA072D2@freeswitch.org>

When we convert them from 48k we can lower the vol a bit more we are  
already doing it slightly.

/b

On Dec 22, 2008, at 3:38 PM, Andy Spitzer wrote:

> I'm not sure if FS can do anything about this, it may be a Polycom  
> issue, but is there a way to apply gain (negative in this case!)  
> during the transcoding process so the percieved volume of the  
> prompts is the same no matter which codec is selected?



From woof at nortel.com  Mon Dec 22 13:55:05 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 22 Dec 2008 16:55:05 -0500
Subject: [Freeswitch-users] Extra loud prompts when transcoded from
 L16@8000 to G.722
In-Reply-To: <3A489B44-DFD2-4D1F-899E-9E7CFEA072D2@freeswitch.org>
References: <op.umk2emi1ssieo1@woof.us.nortel.com>
	<3A489B44-DFD2-4D1F-899E-9E7CFEA072D2@freeswitch.org>
Message-ID: <op.umk253ebssieo1@woof.us.nortel.com>

Woof!

On Mon, 22 Dec 2008 16:46:14 -0500, Brian West <brian at freeswitch.org> wrote:

> When we convert them from 48k we can lower the vol a bit more we are
> already doing it slightly.
> 

The prompts we are using aren't from the FS set.  It's not a matter of adjusting the prompts, they've work fine for G.711 for years now--it's that when real-time transcoded by FS to G.722 the volume is loud.

Also, consider a call that comes in via G.711 and records a message, saved as L16 at 8000 in a .wav file.  Now play that recording back over G.722.  It's way louder than if played back over G.711.  So depending on which phone you pick up your messages on, the difference in percieved volume is quite dramatic.


--Woof!


From gilbertandrew at me.com  Mon Dec 22 13:56:07 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Mon, 22 Dec 2008 16:56:07 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
 work?
In-Reply-To: <8CB327545F492AF-928-221@WEBMAIL-MA02.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
	<8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
	<191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
	<8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
	<8CB327545F492AF-928-221@WEBMAIL-MA02.sysops.aol.com>
Message-ID: <37FBF657-259F-4D02-B295-05CEEDA7561F@me.com>

Mark,

This is great, a second box should help, you might want to go back to  
stock setups as well (ie none of the prophecy port changes, etc). This  
will help eliminate configuration error as an issue.

Anthony, will certainly ping you guys when I start testing. Thanks.

Andy



On Dec 22, 2008, at 12:53 PM, mszlazak at aol.com wrote:

> Hi Andy and Anthony.
>
> Thanks Anthony for elaborating more and I'll attempt using another  
> IP on the same box as well.
>
> Also, Prophecy support has asked me first to put one application on  
> a separate box and then get some wireshark data so I'll attempt that  
> also.
>
> Andy, I didn't want to bother you given all those things you had to  
> deal with. Welcome back.
>
> I explored the VMware idea before but was warned that it would not  
> work well with an ASR. This advice came from the Trixbox forums,  
> LumenVox, FreeSwitch and Voxeo.
>
> I understand that what I'm doing goes against the grain (i.e. voip)  
> but frankly my target market really doesn't want anything to do with  
> voip or even internet connectivity from their businesses. Plus there  
> are other issues.
>
> I'll let you know how it goes.
>
> Happy holidays.
>
> Mark.
>
> -----Original Message-----
> From: Andrew Gilbert <gilbertandrew at me.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 22 Dec 2008 7:49 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
> Proxy Media work?
>
> Mark,
>
> Sorry I haven't had much time to help with this either.
>
> But Anthony is offering good advice here. You are either going to  
> have to work out what is going on at SIP/SDP/RTP level through logs  
> and wireshark, or opt fo r a separate ip space. Another option  
> (besides virtual ips) is VMWare or VirtualBox, although VMWare is  
> probably easier to setup and bridge naturally to your host.
>
> Vm's are just so easy anymore and it definitely seems like you are  
> going against the grain right now.
>
> Also - realizing you got here because of the need for ASR. I do have  
> the Lumenvox license, and I was able to compile the module out of  
> SVN. I have not tested anything yet. If things go well I should have  
> some time after the 25th for this. My goal would be to get pizza or  
> something akin to work.
>
> Andy
>
>
> On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:
>
>> I don't really know what your problem is.  I just saw you ask 3  
>> times for help and tried to offer a suggestion.
>> if you start FS with TPORT_LOG=1 you can see all the sip messages  
>> in the console and you could
>> also run wireshark to look at a packet capture.
>>
>> If you use the same IP for media on the same box for 3 programs at  
>> once you may end up with 2 applictions choosing the same media port  
>> etc.
>>
>> It's just a good practice to run every voip program on it's own IP.
>>
>>
>>
>>
>> On Mon, Dec 22, 2008 at 12:44 AM, <mszlazak at aol.com> wrote:
>> Hi Anthony,
>>
>> I actually suggested adding IP's to a Voxeo-Prophecy support person  
>> before but they thought that could be problematic. I went along  
>> with the earlier warning but now you have suggested it again. What  
>> makes everything on the same box tricky?
>>
>> Also, the thing that surprises me a bit is that bypass-media works  
>> but proxy-media or the default doesn't.
>>
>> Would you be kind enough to elaborate.
>>
>> Thanks. Mark.
>>
>>
>>
>> -----Original Message-----
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Sun, 21 Dec 2008 2:49 pm
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
>> Proxy Media work?
>>
>> Try adding more ip to your box and give each thing it's own  
>> dedicated virtual IP.
>> Doing everything on the same box can be tricky.
>>
>>
>> On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:
>> With the firewall ON or OFF the problem still remains.
>>
>> I've tried 3 different set-ups in a dial plan extension.
>>
>> 1. With  only <action application="set" data="proxy_media=true"/>  
>> before bridging.
>>
>> 2. With only  <action application="set" data="bypass_media=true"/>  
>> before bridging.
>>
>> 3. Neither of the above in the extension.
>>
>> Only 2 with "bypass-media=true" gets the audio across endpoints.
>>
>> Help :-)
>>
>>
>> -----Original Message-----
>> From: mszlazak at aol.com
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Fri, 19 Dec 2008 11:30 am
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
>> Proxy Media work?
>>
>>  With the firewall ON or OFF the problem still remains.
>>
>> I've tried 3 different set-ups in a dial plan extension.
>>
>> 1. With  only <action application="set" data="proxy_media=true"/>  
>> before bridging.
>>
>> 2. With only  <action application="set" data="bypass_media=true"/>  
>> before bridging.
>>
>> 3. Neither of the above in the extension.
>>
>> Only 2 with "proxy-media=true" gets the audio across endpoints.
>>
>> Help :-)
>>
>>
>>
>>
>>
>> 0A
>>
>>
>> -----Original Message-----
>> From: Michael Jerris <mike at jerris.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Fri, 19 Dec 2008 7:49 am
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
>> Proxy Media work?
>>
>> It gives me the impression there is something wrong with your  
>> firewall running on the box.
>>
>> Mike
>>
>> On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>>
>>> I find it strange that I can have to endpoints get audio went  
>>> using bypass media mode but the audio fails to go across endpoints  
>>> if I use proxy media mode.
>>> I'm trying to pass audio "internally" on the same machine between  
>>> endpoints and have be advis ed that a reason the audio may fail to  
>>> be passed is because there is some RTP timing and IP address/port  
>>> issues.
>>> However, FS has no problem "connecting" ports if i change the mode  
>>> to bypass media. This gives me the impression that something is  
>>> wrong with FS proxy media mode.
>>> Any comments?
>>>
>>> Listen to 350+ music, sports, & news radio stations ? including  
>>> songs for the holidays ? FREE while you browse. Start Listening  
>>> Now!
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch. org
>>
>> =
>> _______________________________________________
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Listen to 350+ music, sports, & news radio stations ? including  
>> songs for the holidays ? FREE while you browse. Start Listening  
>> Now!
>>  _______________________________________________
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>> Listen to 350+ music, sports, & news radio stations ? including  
>> songs for the holidays ? FREE while you browse. Start Listening  
>> Now!
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>  _______________________________________________
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>> Freeswitch-users at lists
>>
>> .freeswitch.org
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> Listen to 350+ music, sports, & news radio stations ? including  
>> songs for the holidays ? FREE while you browse. Start Listening Now!
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> =
> _____________________
> __________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mszlazak at aol.com  Mon Dec 22 14:35:38 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 22 Dec 2008 17:35:38 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <37FBF657-259F-4D02-B295-05CEEDA7561F@me.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com><B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com><8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com><8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com><191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com><8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com><191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com><8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com><8CB327545F492AF-928-221@WEBMAIL-MA02.sysops.aol.com>
	<37FBF657-259F-4D02-B295-05CEEDA7561F@me.com>
Message-ID: <8CB329CBF686287-C78-317@WEBMAIL-DG02.sim.aol.com>


 Hi Andy,

Yes, it works and I kept my port changes. 

I tried the set-up in both "bypass-media" mode and what I've called "default" mode (i.e. no "bypass-media" nor "proxy-media" settings). Didn't try proxy-mode since the default worked.

The team at Prophecy is almost 100% certain as to what the problem is and which ports are involved. I've sent them the pcap and Prophecy log files.

Also, I was having background "crackling" noise in "bypass-mode" whether I ran FS in the box that it shared with Prophecy or if FS was running from the other box that didn't have Prophecy.

However, when I ran FS from the box without Prophecy in "default" mode then the audio totally cleared up??


 
Mark.


 

-----Original Message-----
From: Andrew Gilbert <gilbertandrew at me.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 22 Dec 2008 1:56 pm
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?










Mark,



This is great, a second box should help, you might want to go back to stock setups as well (ie none of the prophecy port changes, etc). This will help eliminate configuration error as an issue.



Anthony, will certainly ping you guys when I start testing. Thanks.




Andy









On Dec 22, 2008, at 12:53 PM, mszlazak at aol.com wrote:



 Hi Andy and Anthony.
 
 Thanks Anthony for elaborating more and I'll attempt using another IP on the same box as well. 
 
 Also, Prophecy support has asked me first to put one
 application on a separate box and then get some wireshark data so I'll attempt that also.
 
 Andy, I didn't want to bother you given all those things you had to deal with. Welcome back.
 
 I explored the VMware idea before but was warned that it would not work well with an ASR. This advice came from the Trixbox forums, LumenVox, FreeSwitch and Voxeo.
 
 I understand that what I'm doing goes against the grain (i.e. voip) but frankly my target market really doesn't want anything to do with voip or even internet connectivity from their businesses. Plus there are other issues.
 
 I'll let you know how it goes.
 
 Happy holidays.
 
 Mark.
 
 
 
 
 -----Original Message-----
 From: Andrew Gilbert <gilbertandrew at me.com>
 To: freeswitch-users at lists.freeswitch.org
 Sent: Mon, 22 Dec 2008 7:49 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 Mark, 

 
 
Sorry I haven't had much time to help with this either.
 

 
 
But Anthony is offering good advice here.?You are either going to have to work out what is going on at SIP/SDP/RTP level through logs and wireshark, or opt fo r a separate ip space. Another option (besides virtual ips) is VMWare or VirtualBox, although VMWare is probably easier to setup and bridge naturally to your host.
 

 
 
Vm's are just so easy anymore and it definitely seems like you are going against the grain right now.
 

 
 
Also - realizing you got here because of the ne
ed for ASR. I do have the Lumenvox license, and I was able to compile the module out of SVN. I have not tested anything yet. If things go well I should have some time after the 25th for this. My goal would be to get pizza or something akin to work.
 

 
 
Andy
 

 
 

 
 
On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:
 

I don't really know what your problem is.? I just saw you ask 3 times for help and tried to offer a suggestion.
 if you start FS with TPORT_LOG=1 you can see all the sip messages in the console and you could 
 also run wireshark to look at a packet capture.
 
 If you use the same IP for media on the same box for 3 programs at once you may end up with 2 applictions choosing the same media port etc.
 
 It's just a good practice to run every voip program on it's own IP.
 
 
 
 
 
On Mon, Dec 22, 2008 at 12:44 AM,  <mszlazak at aol.com> wrote:
 
 
 Hi Anthony,
 
 I actually suggested adding IP's to a Voxeo-Prophecy support person before but they thought that could be problematic. I went along with the earlier warning but now you have suggested it again. What makes everything on the same box tricky? 
 
 Also, the thing that surprises me a bit is that bypass-media works but proxy-media or the default doesn't. 
 
 Would you be kind enough to elaborate.
 
 Thanks. Mark.
 
 
 
 
 
 
 
 
 
 -----Original Message-----
 From: Anthony Minessale <anthony.minessale
@gmail.com>
 To: freeswitch-users at lists.freeswitch.org
 
 
 

 
 Sent: Sun, 21 Dec 2008 2:49 pm
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 Try adding more ip to your box and give each thing it's own dedicated virtual IP.
 Doing everything on the same box can be tricky.
 
 
 
On Sat, Dec 20, 2008 at 2:17 AM,  <mszlazak at aol.com> wrote:
 
 
With the firewall ON or OFF the problem still remains.
 
 I've tried 3 different set-ups in a dial plan extension.
 
 1. With? only <action application="set" data="proxy_media=true"/> before bridging.
 ??? ??? ??? 
 2. With only? <action application="set" data="bypass_media=true"/> before bridging.
 
 3. Neither of the above in the extension.
 
 
 Only 2 with "bypass-media=true" gets the audio across endpoints.
 
 Help :-)
  
 
 
 
 
 
 
 -----Original Message-----
 From: mszlazak at aol.com
 To: freeswitch-users at lists.freeswitch.org
 
 
 Sent: Fri, 19 Dec 2008 11:30 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 
 
 
?With the firewall ON or OFF the problem still remains.
 
 I've tried 3 different set-ups in a dial plan extension.
 
 1. With? only <action application="set" data="proxy_media=true"/> before bridging.
 ??? ??? ??? 
 2. With only? <action application="set" data="bypass_media=true"/> before bridging.
 
 3.
 Neither of the above in the extension.
 
 Only 2 with "proxy-media=true" gets the audio across endpoints.
 
 Help :-)
 
 
 ? 
 
 
 
 
 0A 
 

 
 
 
 
 
 
 
 -----Original Message-----
 From: Michael Jerris <mike at jerris.com>
 To: freeswitch-users at lists.freeswitch.org
 Sent: Fri, 19 Dec 2008 7:49 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 It gives me the impression there is something wrong with your firewall running on the box. 

 
 
Mike
 

 
 
 
 
On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
 
 
I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.
  I'm trying to pass audio "internally" on the same machine between endpoints and have be advis ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 
  However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.
  Any comments?? 
   

 

Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
  _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch. org
 

 
 
 = 
  
 
_______________________________________________















Freeswitch-users mailing list















Freeswitch-users at lists.freeswitch.org














http://lists.freeswitch.org/mailman/listinfo/freeswitch-users















UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users















http://www.freeswitch.org
















 
  

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 
 

 
   
 
_______________________________________________













Freeswitch-users mailing list







Freeswitch-users at lists.freeswitch.org







http://lists.freeswitch.org/mailman/listinfo/freeswitch-users







UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users







http://www.freeswitch.org








 
   

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 

 
 

 --20
 Anthony Minessale II
 
 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 
 AIM: anthm
 MSN:anthony_minessale at hotmail.com
 GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
 IRC: irc.freenode.net #freeswitch
 
 FreeSWITCH Developer Conference
 sip:888 at conference.freeswitch.org
 iax:guest at conference.freeswitch.org/888
 googletalk:conf+888 at conference.freeswitch.org
 pstn:213-799-1400
 
 
 
_______________________________________________



Freeswitch-users mailing list



Freeswitch-users at lists

.freeswitch.org



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



http://www.freeswitch.org




 
 

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 



 
 

 -- 
 Anthony Minessale II
 
 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 
 AIM: anthm
 MSN:anthony_minessale at hotmail.com
 GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
 IRC: irc.freenode.net #freeswitch
 
 FreeSWITCH Developer Conference
 sip:888 at conference.freeswitch.org
 iax:guest at conference.freeswitch.org/888
 googletalk:conf+888 at conference.freeswitch.org
 pstn:213-799-1400
 _______________________________________________
 Freeswitch-users mailing list
 Freeswitch
-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 
 
 =   
 
_____________________
__________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org


 
  



Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 _______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From jaugenstine at gmail.com  Mon Dec 22 14:51:43 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Mon, 22 Dec 2008 14:51:43 -0800
Subject: [Freeswitch-users] mod_java.so load issue
In-Reply-To: <F009F62E-F171-449D-9B70-4E1090B1DBF0@jerris.com>
References: <207e7a5e0812201104l6280ba16g265486f750f10604@mail.gmail.com>
	<F009F62E-F171-449D-9B70-4E1090B1DBF0@jerris.com>
Message-ID: <207e7a5e0812221451r73decf09j5e6806775290fd5e@mail.gmail.com>

Mike

Could the problem be related to the gcc/g++ version.  The one installed is:
g++ (GCC) 4.3.0 20080428 (Red Hat 4.3.0-8)
I have been trying to look at differences between this server and others
servers I have installed on.  So far, that is the only significant
difference I can identify.  Previously, the gcc version I have built with is
4.0.1.

Jonathan

On Sat, Dec 20, 2008 at 11:33 AM, Michael Jerris <mike at jerris.com> wrote:

> I would suggest cleaning and rebuilding the module.  If that doesn't
> work could we arrange access to the box so I can take a look?
>
> Mike
>
> On Dec 20, 2008, at 2:04 PM, "jonathan augenstine" <jaugenstine at gmail.com
>  > wrote:
>
> > I am installing Freeswitch on Fedora.  I was building/installing the
> > mod_java.so module and I encountered the following load issue:
> >
> > 2008-12-20 10:34:58 [CRIT] switch_loadable_module.c:839
> > switch_loadable_module_load_file() Error Loading module /usr/local/
> > freeswitch/mod/mod_java.so
> > **/usr/local/freeswitch/mod/mod_java.so: invalid ELF header**
> >
> > Is this a build issue?  I am assuming maybe there is a g++ option
> > that is set incorrectly but my searches on Google and looking at gcc
> > docs have not provided a solution.
> >
> > Jonathan
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Mon Dec 22 15:01:02 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 23 Dec 2008 10:01:02 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <38E75A1D-7391-46A8-BD9C-1C851A019625@freeswitch.org>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<38E75A1D-7391-46A8-BD9C-1C851A019625@freeswitch.org>
Message-ID: <20081222230102.GA7220@jdc.jasonjgw.net>

On Mon, Dec 15, 2008 at 08:45:49AM -0600, Brian West wrote:
> Are you using SVN trunk?  This has been fixed already as far as I  
> remember!!

Just to close out this issue for now, it is possible that the FreeSWITCH
version I was running when this problem occurred included object files from an
earlier build; I ran "make clean" before compiling it, but not "debuild
clean", which cleans all of the libraries as well.

It is also clear that a Hurricane Electric router on the path to New York has
been experiencing issues, which account for some of my IPv6 problems, as the
destination of the calls was connected via tunnel to the New York point of
presence.



From msc at freeswitch.org  Mon Dec 22 15:09:43 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 15:09:43 -0800
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081222230102.GA7220@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<38E75A1D-7391-46A8-BD9C-1C851A019625@freeswitch.org>
	<20081222230102.GA7220@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812221509k1412a552i37c1a2c040603df@mail.gmail.com>

Thanks for keeping us updated! It's a lot nicer than, "I wonder whatever
happened to Jason and his IPv6 issue?"
:)

-MC

On Mon, Dec 22, 2008 at 3:01 PM, Jason White <jason at jasonjgw.net> wrote:

> On Mon, Dec 15, 2008 at 08:45:49AM -0600, Brian West wrote:
> > Are you using SVN trunk?  This has been fixed already as far as I
> > remember!!
>
> Just to close out this issue for now, it is possible that the FreeSWITCH
> version I was running when this problem occurred included object files from
> an
> earlier build; I ran "make clean" before compiling it, but not "debuild
> clean", which cleans all of the libraries as well.
>
> It is also clear that a Hurricane Electric router on the path to New York
> has
> been experiencing issues, which account for some of my IPv6 problems, as
> the
> destination of the calls was connected via tunnel to the New York point of
> presence.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Dec 22 15:22:30 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 15:22:30 -0800
Subject: [Freeswitch-users] Freeswitch/Sofia configuration problem
In-Reply-To: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>
References: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>
Message-ID: <87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>

Are you using the default config? If you've made any changes at all we'd
need to know about them. Also, can you turn on SIP trace so that we can see
exactly what is coming and going?
Start FS like this:
TPORT_LOG=1 ./freeswitch
Press F8 to put the console in debug mode
then capture the output while you observe the bad behavior
Please put all that, plus any config changes, into a pastebin:
pastebin.freeswitch.org

I'm sure there are people around here who can help you figure out what is
going on.
-MC

On Mon, Dec 22, 2008 at 8:20 AM, Laurent Fabre
<Laurent.Fabre at kirranet.com>wrote:

>
> Hello,
>
> I've been trying to figure out for a few days why my freeswitch instance
> suddenly become insensitive to SIP packets without any warning.
>
> What usually happen is the following :
>
> 1) start just fine in foreground mode and no errors
> 2) wait anywhere between 2 seconds and 20 minutes
> 3) Sofia suddenly decide to reload everything for some reason
> 4) Sofia start processing SIP packets
> 5) work for an hour or so
> 6) Sofia suddenly decide to reload everything for some reason
> 7) become unresponsive again
> 8) goto 2
>
> Both interfaces have public IP addresses assigned in a static manner (no
> DHCP).
>
> I can see the SIP UDP & TCP requests comming from the phones on several
> sites on the wire.
> The SIP TCP requests get RST in reply which is mean :(
>
> There was a point in my setup where it would not happen but since I'm new
> to freeswitch I'm having an hard time backtracking.
>
> I was wondering if iproute/tc and iptables were the culprits but I flushed
> everything (even rebooted without loading the rules) and it still doesn't
> work.
>
> I thought some database was corrupt so I shutdown'd freeswitch and delete
> his db folder, no effect.
>
> My server runs Debian 4.0etch for amd64, built freeswitch from SVN trunk.
>
> Any pointers, help, cure against headaches would be great :)
>
> Regards,
>
> Laurent
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Dec 22 16:01:07 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 18:01:07 -0600
Subject: [Freeswitch-users] Freeswitch/Sofia configuration problem
In-Reply-To: <87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>
References: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>
	<87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>
Message-ID: <D8AB2778-AEF3-48F5-9B95-DC057AA3C581@freeswitch.org>

The problem is FS thinks your IP has changed... you need <param  
name="auto-restart" value="false"/> in sofia.conf.xml

/b

On Dec 22, 2008, at 5:22 PM, Michael Collins wrote:

> Are you using the default config? If you've made any changes at all  
> we'd need to know about them. Also, can you turn on SIP trace so  
> that we can see exactly what is coming and going?
> Start FS like this:
> TPORT_LOG=1 ./freeswitch
> Press F8 to put the console in debug mode
> then capture the output while you observe the bad behavior
> Please put all that, plus any config changes, into a pastebin:  
> pastebin.freeswitch.org
>
> I'm sure there are people around here who can help you figure out  
> what is going on.
> -MC

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From brian at freeswitch.org  Mon Dec 22 16:51:26 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 18:51:26 -0600
Subject: [Freeswitch-users] Extra loud prompts when transcoded from
	L16@8000 to G.722
In-Reply-To: <op.umk253ebssieo1@woof.us.nortel.com>
References: <op.umk2emi1ssieo1@woof.us.nortel.com>
	<3A489B44-DFD2-4D1F-899E-9E7CFEA072D2@freeswitch.org>
	<op.umk253ebssieo1@woof.us.nortel.com>
Message-ID: <7E0C2F90-1AA9-4A15-BAEB-C4A14DA9A932@freeswitch.org>

can you provide me sound file samples?

/b

On Dec 22, 2008, at 3:55 PM, Andy Spitzer wrote:

> Woof!
>
> On Mon, 22 Dec 2008 16:46:14 -0500, Brian West  
> <brian at freeswitch.org> wrote:
>
>> When we convert them from 48k we can lower the vol a bit more we are
>> already doing it slightly.
>>
>
> The prompts we are using aren't from the FS set.  It's not a matter  
> of adjusting the prompts, they've work fine for G.711 for years now-- 
> it's that when real-time transcoded by FS to G.722 the volume is loud.
>
> Also, consider a call that comes in via G.711 and records a message,  
> saved as L16 at 8000 in a .wav file.  Now play that recording back over  
> G.722.  It's way louder than if played back over G.711.  So  
> depending on which phone you pick up your messages on, the  
> difference in percieved volume is quite dramatic.
>
>
> --Woof!

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From john at feith.com  Mon Dec 22 17:10:00 2008
From: john at feith.com (John Wehle)
Date: Mon, 22 Dec 2008 20:10:00 -0500 (EST)
Subject: [Freeswitch-users]  another switch_ivr_set_user() can't find user
Message-ID: <200812230110.mBN1A0t6004432@jwlab.FEITH.COM>

I'm interested in moving some VoIP phones I'm playing with to a different
set of network numbers for various internal reasons.  However, I'm not
looking for multiple FreeSWITCH domains ... I just want one.  Here's what
I did:

  1) On the FreeSWITCH box created a logical network interface and assigned
     it a number from the new set of network numbers.

  2) Changed rtp-ip and sip-ip in internal.xml to use the number assigned
     to the logical network interface.

  3) Per the FAQ set force-register-domain and force-register-db-domain in
     internal.xml to the value for domain in vars.xml (i.e. the main local
     ip v4 address of the FreeSWITCH machine).

  4) Changed the VoIP phone to use a number from the new set.

When I place a call from the VoIP phone FreeSWITCH complains:

  [WARNING] switch_ivr.c:1941 switch_ivr_set_user() can't find user
    [default at 192.168.14.10]
 
where 192.168.14.10 is the number assigned to the logical interface,
however the call goes through / everything seems to work.  I'm able
to place a call to the VoIP phone from openzap without any complaints.

Suggestions / pointers regarding the warning and how to make it happy?

-- John
-------------------------------------------------------------------------
|   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
|    John Wehle    |     Fax: 1-215-540-5495  |                         |
-------------------------------------------------------------------------



From pieter_eduard at biznetnetworks.com  Mon Dec 22 20:00:26 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Tue, 23 Dec 2008 11:00:26 +0700
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>
References: <494F93EA.4080608@biznetnetworks.com>
	<CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>
Message-ID: <4950625A.90100@biznetnetworks.com>

Hi Brian,

I already update the fs to FreeSWITCH Version 1.0.trunk (10906M), and 
the problem still persist .
I already attach the debug log using "console loglevel debug" and post 
it at the jira, and I'm coordinating with the provider regarding 
Anthony's comment about the nat / firewall issue so i hope i could get 
some response from them.

Thank you,

-Pieter-

Brian West wrote:
> I also need you to do this call again with "console loglevel debug" on
> and post it attached to the jira and not inline on the comments please.
>
> /b
> On Dec 22, 2008, at 7:19 AM, Pieter Eduard wrote:
>
>   
>> Hi,
>>
>> I have an enum number, if I call the number from any ip extension ( i
>> use default enum.conf that points to e164.arpa)  then the call goes
>> well
>> to my ATA that registers to my fs box,
>> but if i try to call the number from PLMN, i get the ring at my ATA
>> and
>> if i pick it up, there's no sound.
>>
>> here's my public.xml config :
>>
>> <extension name="enum">
>>        <condition field="destination_number" expression="^12345678">
>>        <action application="set" data="proxy_media=true"/>
>>        <action application="bridge" data="sofia/default/
>> 1000%1.2.3.4"/>
>>        </condition>
>>    </extension>
>>
>> For more detailed debug log, i already submit it on jira :
>> http://jira.freeswitch.org/browse/MODAPP-186
>>
>>
>> regards,
>>
>> -Pieter-
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From jason at jasonjgw.net  Mon Dec 22 20:12:37 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 23 Dec 2008 15:12:37 +1100
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find	user
In-Reply-To: <200812230110.mBN1A0t6004432@jwlab.FEITH.COM>
References: <200812230110.mBN1A0t6004432@jwlab.FEITH.COM>
Message-ID: <20081223041237.GA19348@jdc.jasonjgw.net>

On Mon, Dec 22, 2008 at 08:10:00PM -0500, John Wehle wrote:
> I'm interested in moving some VoIP phones I'm playing with to a different
> set of network numbers for various internal reasons.  However, I'm not
> looking for multiple FreeSWITCH domains ... I just want one.  Here's what
> I did:
> 
>   1) On the FreeSWITCH box created a logical network interface and assigned
>      it a number from the new set of network numbers.
> 
>   2) Changed rtp-ip and sip-ip in internal.xml to use the number assigned
>      to the logical network interface.
> 
>   3) Per the FAQ set force-register-domain and force-register-db-domain in
>      internal.xml to the value for domain in vars.xml (i.e. the main local
>      ip v4 address of the FreeSWITCH machine).
> 
>   4) Changed the VoIP phone to use a number from the new set.
> 

All fine so far.
> When I place a call from the VoIP phone FreeSWITCH complains:
> 
>   [WARNING] switch_ivr.c:1941 switch_ivr_set_user() can't find user
>     [default at 192.168.14.10]
>  
> where 192.168.14.10 is the number assigned to the logical interface,
> however the call goes through / everything seems to work.  

Somewhere in your dial plan, the set_user application is being called with
the above user and domain as parameter.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user

The logs should show you which extensions are being executed in the dial plan
so you can work out how it reached this point and why it's invoking set_user
there. Basically, work through the logic of your dial plan to find out why
this is happening.

I'm sure others will have more specific advice, but, basically, it has to do
with the details of how your dial plan is configured.



From brian at freeswitch.org  Mon Dec 22 20:13:04 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 22:13:04 -0600
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <4950625A.90100@biznetnetworks.com>
References: <494F93EA.4080608@biznetnetworks.com>
	<CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>
	<4950625A.90100@biznetnetworks.com>
Message-ID: <86D1DBA0-07B3-47A7-9CE4-B474166C5E06@freeswitch.org>

First off you opened a jira and assigned it to yourself.. not exactly  
the best way to get it fixed.. secondly what is a PLMN in this  
context?  ... Thirdly... if you notice the bye is sent three times  
which is a big indication that you have a nat problem since its never  
ACK'ed.  Gather up a pcap and attach it to the bug along with a  
console log on "console loglevel debug" attached... please do not  
paste them inline.

/b

On Dec 22, 2008, at 10:00 PM, Pieter Eduard wrote:

> Hi Brian,
>
> I already update the fs to FreeSWITCH Version 1.0.trunk (10906M),  
> and the problem still persist .
> I already attach the debug log using "console loglevel debug" and  
> post it at the jira, and I'm coordinating with the provider  
> regarding Anthony's comment about the nat / firewall issue so i hope  
> i could get some response from them.
>
> Thank you,
>
> -Pieter-



From pieter_eduard at biznetnetworks.com  Mon Dec 22 20:27:37 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Tue, 23 Dec 2008 11:27:37 +0700
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <86D1DBA0-07B3-47A7-9CE4-B474166C5E06@freeswitch.org>
References: <494F93EA.4080608@biznetnetworks.com>	<CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>	<4950625A.90100@biznetnetworks.com>
	<86D1DBA0-07B3-47A7-9CE4-B474166C5E06@freeswitch.org>
Message-ID: <495068B9.5070606@biznetnetworks.com>

Hi Brian,

sorry about the jira, i closed the too much inline log and gonna open a 
new one after i have some results along with the wireshark dump test.

thank you,

-Pieter-

Brian West wrote:
> First off you opened a jira and assigned it to yourself.. not exactly
> the best way to get it fixed.. secondly what is a PLMN in this
> context?  ... Thirdly... if you notice the bye is sent three times
> which is a big indication that you have a nat problem since its never
> ACK'ed.  Gather up a pcap and attach it to the bug along with a
> console log on "console loglevel debug" attached... please do not
> paste them inline.
>
> /b
>
> On Dec 22, 2008, at 10:00 PM, Pieter Eduard wrote:
>
>   
>> Hi Brian,
>>
>> I already update the fs to FreeSWITCH Version 1.0.trunk (10906M),
>> and the problem still persist .
>> I already attach the debug log using "console loglevel debug" and
>> post it at the jira, and I'm coordinating with the provider
>> regarding Anthony's comment about the nat / firewall issue so i hope
>> i could get some response from them.
>>
>> Thank you,
>>
>> -Pieter-
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From carole.olivier at enst.fr  Mon Dec 22 23:03:29 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Mon, 22 Dec 2008 23:03:29 -0800 (PST)
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
References: <21127913.post@talk.nabble.com>
	<2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
Message-ID: <21140461.post@talk.nabble.com>


Hello,

When I do a "show channels" in the cli the channels to the speakers are
listed even if the speakers have stopped transmitting. If I call the
speakers again freeswitch create new channels. If I do a "show channels"
again I can see the old and new ones. If I can keep doing this, each time
new channels are created while the old ones are still there.
I have noticed after 5 minutes the channels that are not used anymore close.
I believed there was a kind of timeout to detect the channels that are not
in use.

What I would like to know is if there is a way to close from these channels
the dialplan .

Thanks,
Carole




Brian West-3 wrote:
> 
> What do you mean they close (hangup) after the 5 minute timeout?
> 
> /b
> 
> On Dec 22, 2008, at 7:35 AM, Carole O. wrote:
> 
>> 1021 and 1022 are the speakers.
>> At the end of the announcement, since there is no noise anymore, the
>> speakers stop listening but they do not send any messages to tell  
>> Freeswitch
>> it can close the opened channels. The channels are closed only after a
>> timeout of 5 minutes.
>>
>> Does anybody know how I could force freeswitch to close all the  
>> channels
>> after the announcement? I have seen there is the application  
>> sched_hangup
>> but when I used it it only closes the channel to the caller and not  
>> the
>> other ones.
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21140461.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From odermann at googlemail.com  Tue Dec 23 00:49:38 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 09:49:38 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
Message-ID: <5e414ed0812230049g16928608md8b7e498cf7b8dce@mail.gmail.com>

hi,

i am quite new to freeswitch and now i finally have fs up and running
as i want with my php scripts to handle the calls.

now that i want to start the service in the near future, i would like
to test the performance of the whole system and the reliability.

what are your experiences and ideas, how i could do it the best way?
and what do i have to look for?

i startet to test with sipp, which seems to be a useful tool, although
i am not sure, how to read the results.

in the switch.conf i set "max-sessions" to 11000 and
"sessions-per-second" to 2000.

in the console i entered: sipp -sn uac xx.xx.xx.xx -s 123456 -r 20 -l
1000 -d 4000 -rtp_echo

the results are as follow:

20.0(4000 ms)/1.000s   5061      51.18 s         1023  xx.xx.xx.xx:5060(UDP)

  20 new calls during 1.001 s period     9 ms scheduler resolution
  807 calls (limit 1000)                 Peak was 807 calls, after 51 s
  0 Running, 807 Paused, 0 Woken up
  0 out-of-call msg (discarded)
  1 open sockets
  75989 Total echo RTP pckts 1st stream  481.979 last period RTP rate (kB/s)
  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)

                                 Messages  Retrans   Timeout   Unexpected-Msg
      INVITE ---------->         1023      0         0
         100 <----------         1023      0                   0
         180 <----------         0         0                   0
         183 <----------         249       0                   0
         200 <----------  E-RTD1 249       0                   0
         ACK ---------->         249       0
       Pause [      0ms]         249                           0
         BYE ---------->         216       0         0
         200 <----------         216       0                   0



  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:072              | 00:08:30:106
  Call Rate              |    0.000 cps              |    4.166 cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |        0
  OutGoing call created  |        0                  |     2125
  Total Call created     |                           |     2125
  Current Call           |        0                  |
-------------------------+---------------------------+--------------------------
  Successful call        |        9                  |     2125
  Failed call            |        0                  |        0
-------------------------+---------------------------+--------------------------
  Response Time 1        | 00:00:00:000              | 00:02:43:140
  Call Length            | 00:04:04:169              | 00:02:47:149



i do not get any errors. but i think that it is strange, that there
are so little calls running at the same time. when i try to call the
same number as sipp does, i have to wait a very long time, till the
call is beeing answered.

why are there so many paused calls waiting for fs to take care of?

is this normal or are there some problems, which i have to take care of?

i want to add, that i am using socket outbound with a number of
php-scripts, which do a lot of call- and dialplan-handling. i am not
using any xml. therefore i am sure, that the results are worse, that
they would be with normal xml dialplans.


thanks for your help and tipps.
dannis


From odermann at googlemail.com  Tue Dec 23 00:52:43 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 09:52:43 +0100
Subject: [Freeswitch-users] Memory question
Message-ID: <5e414ed0812230052n71705a25s6d55e694d609b148@mail.gmail.com>

after doing some testing with fs, i can see in the console, when
entering "top", that fs uses 9.9% of the memory.

when i do some more calls, the used memory will raise - the memory
will not beeing released, till i do a restart of fs.

is this a normal behavior or do i have some problems?


thanks
dennis


From krice at suspicious.org  Tue Dec 23 01:01:44 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:01:44 -0600
Subject: [Freeswitch-users] Memory question
In-Reply-To: <5e414ed0812230052n71705a25s6d55e694d609b148@mail.gmail.com>
Message-ID: <C5760518.4B5C0%krice@suspicious.org>

This is normal behavior... FS allocates memory into pools and re-uses that
same memory over and over... It is quite normal to see memory usage increase
as usage of FS increases to a point where it levels off for that load...  As
the loa decreases memory is not released but used for later when loading
increases again


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 09:52:43 +0100
> To: <Freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Memory question
> 
> after doing some testing with fs, i can see in the console, when
> entering "top", that fs uses 9.9% of the memory.
> 
> when i do some more calls, the used memory will raise - the memory
> will not beeing released, till i do a restart of fs.
> 
> is this a normal behavior or do i have some problems?
> 
> 
> thanks
> dennis
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From krice at suspicious.org  Tue Dec 23 01:03:25 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:03:25 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230049g16928608md8b7e498cf7b8dce@mail.gmail.com>
Message-ID: <C576057D.4B5C2%krice@suspicious.org>

Freeswitch can handle a large volume of call... I suggest you review your
configs to make sure you don't have any of the default or arbitrary other
limits in there... We routinely run > 1500 concurrent calls on dual quad
core hardware at call rates far above what you tested at

Ken


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 09:49:38 +0100
> To: <Freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Performance testing: FS and own App?
> 
> hi,
> 
> i am quite new to freeswitch and now i finally have fs up and running
> as i want with my php scripts to handle the calls.
> 
> now that i want to start the service in the near future, i would like
> to test the performance of the whole system and the reliability.
> 
> what are your experiences and ideas, how i could do it the best way?
> and what do i have to look for?
> 
> i startet to test with sipp, which seems to be a useful tool, although
> i am not sure, how to read the results.
> 
> in the switch.conf i set "max-sessions" to 11000 and
> "sessions-per-second" to 2000.
> 
> in the console i entered: sipp -sn uac xx.xx.xx.xx -s 123456 -r 20 -l
> 1000 -d 4000 -rtp_echo
> 
> the results are as follow:
> 
> 20.0(4000 ms)/1.000s   5061      51.18 s         1023  xx.xx.xx.xx:5060(UDP)
> 
>   20 new calls during 1.001 s period     9 ms scheduler resolution
>   807 calls (limit 1000)                 Peak was 807 calls, after 51 s
>   0 Running, 807 Paused, 0 Woken up
>   0 out-of-call msg (discarded)
>   1 open sockets
>   75989 Total echo RTP pckts 1st stream  481.979 last period RTP rate (kB/s)
>   0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)
> 
>                                  Messages  Retrans   Timeout   Unexpected-Msg
>       INVITE ---------->         1023      0         0
>          100 <----------         1023      0                   0
>          180 <----------         0         0                   0
>          183 <----------         249       0                   0
>          200 <----------  E-RTD1 249       0                   0
>          ACK ---------->         249       0
>        Pause [      0ms]         249                           0
>          BYE ---------->         216       0         0
>          200 <----------         216       0                   0
> 
> 
> 
>   Counter Name           | Periodic value            | Cumulative value
> -------------------------+---------------------------+------------------------
> --
>   Elapsed Time           | 00:00:00:072              | 00:08:30:106
>   Call Rate              |    0.000 cps              |    4.166 cps
> -------------------------+---------------------------+------------------------
> --
>   Incoming call created  |        0                  |        0
>   OutGoing call created  |        0                  |     2125
>   Total Call created     |                           |     2125
>   Current Call           |        0                  |
> -------------------------+---------------------------+------------------------
> --
>   Successful call        |        9                  |     2125
>   Failed call            |        0                  |        0
> -------------------------+---------------------------+------------------------
> --
>   Response Time 1        | 00:00:00:000              | 00:02:43:140
>   Call Length            | 00:04:04:169              | 00:02:47:149
> 
> 
> 
> i do not get any errors. but i think that it is strange, that there
> are so little calls running at the same time. when i try to call the
> same number as sipp does, i have to wait a very long time, till the
> call is beeing answered.
> 
> why are there so many paused calls waiting for fs to take care of?
> 
> is this normal or are there some problems, which i have to take care of?
> 
> i want to add, that i am using socket outbound with a number of
> php-scripts, which do a lot of call- and dialplan-handling. i am not
> using any xml. therefore i am sure, that the results are worse, that
> they would be with normal xml dialplans.
> 
> 
> thanks for your help and tipps.
> dannis
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From odermann at googlemail.com  Tue Dec 23 01:06:04 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 10:06:04 +0100
Subject: [Freeswitch-users] Memory question
In-Reply-To: <C5760518.4B5C0%krice@suspicious.org>
References: <5e414ed0812230052n71705a25s6d55e694d609b148@mail.gmail.com>
	<C5760518.4B5C0%krice@suspicious.org>
Message-ID: <5e414ed0812230106j41acb813xbed5f99727fee92@mail.gmail.com>

thanks for the good explaination and for making me feel better :-)



2008/12/23 Ken Rice <krice at suspicious.org>:
> This is normal behavior... FS allocates memory into pools and re-uses that
> same memory over and over... It is quite normal to see memory usage increase
> as usage of FS increases to a point where it levels off for that load...  As
> the loa decreases memory is not released but used for later when loading
> increases again
>
>
>> From: Dennis <odermann at googlemail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 23 Dec 2008 09:52:43 +0100
>> To: <Freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] Memory question
>>
>> after doing some testing with fs, i can see in the console, when
>> entering "top", that fs uses 9.9% of the memory.
>>
>> when i do some more calls, the used memory will raise - the memory
>> will not beeing released, till i do a restart of fs.
>>
>> is this a normal behavior or do i have some problems?
>>
>>
>> thanks
>> dennis
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From carthick84 at gmail.com  Tue Dec 23 01:10:05 2008
From: carthick84 at gmail.com (B Karthik)
Date: Tue, 23 Dec 2008 14:40:05 +0530
Subject: [Freeswitch-users] Error when building freeswitch on Debian Etch
	64bit.
Message-ID: <c564ccb60812230110j58eb7161rd0c6dc4660d828a5@mail.gmail.com>

Hi,

I am getting the following error when compiling latest Freeswitch with svn
Revision no - 10914 on Debian etch 64bit. Freeswitch version 1.0.1 is
building successfully.

Making all in .
gcc -I/opt/src/freeswitch/src/include
-I/opt/src/freeswitch/libs/libteletone/src -fPIC -Werror -g -ggdb -g -O2
-pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I/opt/src/freeswitch/libs/apr/include
-I/opt/src/freeswitch/libs/apr-util/include -I/opt/src/freeswitch/libs/stfu
-I/opt/src/freeswitch/libs/sqlite -I/opt/src/freeswitch/libs/pcre
-I/opt/src/freeswitch/libs/srtp/include
-I/opt/src/freeswitch/libs/srtp/crypto/include
-I/opt/src/freeswitch/libs/libresample/include -Ilibs/libedit/src
-DSWITCH_HAVE_LIBEDIT -DSWITCH_HAVE_ODBC -Ilibs/libedit/src
-DSWITCH_HAVE_LIBEDIT -g -O2 -Wall -std=c99 -pedantic -o .libs/freeswitch
freeswitch-switch.o  -lm -L/usr/local/lib ./.libs/libfreeswitch.so
libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt -lpthread
libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath
-Wl,/opt/freeswitch/lib
./.libs/libfreeswitch.so: undefined reference to `operator new(unsigned
long)'
./.libs/libfreeswitch.so: undefined reference to `operator delete(void*)'
./.libs/libfreeswitch.so: undefined reference to `__gxx_personality_v0'
./.libs/libfreeswitch.so: undefined reference to `__cxa_pure_virtual'
./.libs/libfreeswitch.so: undefined reference to `vtable for
__cxxabiv1::__class_type_info'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1

Thanks.

B Karthik
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From odermann at googlemail.com  Tue Dec 23 01:31:00 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 10:31:00 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C576057D.4B5C2%krice@suspicious.org>
References: <5e414ed0812230049g16928608md8b7e498cf7b8dce@mail.gmail.com>
	<C576057D.4B5C2%krice@suspicious.org>
Message-ID: <5e414ed0812230131r7c00b1c2ic5b35fdc2d05ceb@mail.gmail.com>

if i do the same test with the 9998, it does not to seem much better:

20.0(4000 ms)/1.000s   5061      68.25 s         1174  xx.xx.xx.xx:5060(UDP)

  0 new calls during 1.008 s period      9 ms scheduler resolution
  710 calls (limit 1000)                 Peak was 782 calls, after 58 s
  0 Running, 710 Paused, 0 Woken up
  0 out-of-call msg (discarded)
  1 open sockets
  595580 Total echo RTP pckts 1st stream 2809.336 last period RTP rate (kB/s)
  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)

                                 Messages  Retrans   Timeout   Unexpected-Msg
      INVITE ---------->         1174      0         0
         100 <----------         1174      0                   0
         180 <----------         0         0                   0
         183 <----------         0         0                   0
         200 <----------  E-RTD1 480       0                   0
         ACK ---------->         480       0
       Pause [      0ms]         480                           0
         BYE ---------->         464       0         0
         200 <----------         464       0                   0

what settings could i review to get more out of the server?

our setup is a new xeon quad core, 4 gb ram and ubuntu 64-bit. we also
entered the ulimit lines and set "manage-presence" to false.


thanks
dennis



2008/12/23 Ken Rice <krice at suspicious.org>:
> Freeswitch can handle a large volume of call... I suggest you review your
> configs to make sure you don't have any of the default or arbitrary other
> limits in there... We routinely run > 1500 concurrent calls on dual quad
> core hardware at call rates far above what you tested at
>
> Ken


From krice at suspicious.org  Tue Dec 23 01:38:02 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:38:02 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230131r7c00b1c2ic5b35fdc2d05ceb@mail.gmail.com>
Message-ID: <C5760D9A.4B5CA%krice@suspicious.org>

There are a number of issues you can be running into... It really depends on
how your app works, what your actual configuration of freeswitch is, disk IO
subsystem, ulimits, etc etc....



> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 10:31:00 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
> 
> if i do the same test with the 9998, it does not to seem much better:
> 
> 20.0(4000 ms)/1.000s   5061      68.25 s         1174  xx.xx.xx.xx:5060(UDP)
> 
>   0 new calls during 1.008 s period      9 ms scheduler resolution
>   710 calls (limit 1000)                 Peak was 782 calls, after 58 s
>   0 Running, 710 Paused, 0 Woken up
>   0 out-of-call msg (discarded)
>   1 open sockets
>   595580 Total echo RTP pckts 1st stream 2809.336 last period RTP rate (kB/s)
>   0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)
> 
>                                  Messages  Retrans   Timeout   Unexpected-Msg
>       INVITE ---------->         1174      0         0
>          100 <----------         1174      0                   0
>          180 <----------         0         0                   0
>          183 <----------         0         0                   0
>          200 <----------  E-RTD1 480       0                   0
>          ACK ---------->         480       0
>        Pause [      0ms]         480                           0
>          BYE ---------->         464       0         0
>          200 <----------         464       0                   0
> 
> what settings could i review to get more out of the server?
> 
> our setup is a new xeon quad core, 4 gb ram and ubuntu 64-bit. we also
> entered the ulimit lines and set "manage-presence" to false.
> 
> 
> thanks
> dennis
> 
> 
> 
> 2008/12/23 Ken Rice <krice at suspicious.org>:
>> Freeswitch can handle a large volume of call... I suggest you review your
>> configs to make sure you don't have any of the default or arbitrary other
>> limits in there... We routinely run > 1500 concurrent calls on dual quad
>> core hardware at call rates far above what you tested at
>> 
>> Ken
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From odermann at googlemail.com  Tue Dec 23 01:43:30 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 10:43:30 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C5760D9A.4B5CA%krice@suspicious.org>
References: <5e414ed0812230131r7c00b1c2ic5b35fdc2d05ceb@mail.gmail.com>
	<C5760D9A.4B5CA%krice@suspicious.org>
Message-ID: <5e414ed0812230143i2f094924s180afea80a66f382@mail.gmail.com>

because the latest result was with the 9998, it can't be out app (at
the moment).

so there are no other typical things or settings i could look for?


2008/12/23 Ken Rice <krice at suspicious.org>:
> There are a number of issues you can be running into... It really depends on
> how your app works, what your actual configuration of freeswitch is, disk IO
> subsystem, ulimits, etc etc....


From krice at suspicious.org  Tue Dec 23 01:48:36 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:48:36 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230143i2f094924s180afea80a66f382@mail.gmail.com>
Message-ID: <C5761014.4B5CE%krice@suspicious.org>

Whats this 9998 to which you refer?


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 10:43:30 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
> 
> because the latest result was with the 9998, it can't be out app (at
> the moment).
> 
> so there are no other typical things or settings i could look for?
> 
> 
> 2008/12/23 Ken Rice <krice at suspicious.org>:
>> There are a number of issues you can be running into... It really depends on
>> how your app works, what your actual configuration of freeswitch is, disk IO
>> subsystem, ulimits, etc etc....
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From odermann at googlemail.com  Tue Dec 23 01:52:09 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 10:52:09 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C5761014.4B5CE%krice@suspicious.org>
References: <5e414ed0812230143i2f094924s180afea80a66f382@mail.gmail.com>
	<C5761014.4B5CE%krice@suspicious.org>
Message-ID: <5e414ed0812230152u78d3ec3ci22961951e8686831@mail.gmail.com>

the 9998 is an extension in the default.xml to test with media flowing
through the line.


2008/12/23 Ken Rice <krice at suspicious.org>:
> Whats this 9998 to which you refer?
>
>
>> From: Dennis <odermann at googlemail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 23 Dec 2008 10:43:30 +0100
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
>>
>> because the latest result was with the 9998, it can't be out app (at
>> the moment).
>>
>> so there are no other typical things or settings i could look for?
>>
>>
>> 2008/12/23 Ken Rice <krice at suspicious.org>:
>>> There are a number of issues you can be running into... It really depends on
>>> how your app works, what your actual configuration of freeswitch is, disk IO
>>> subsystem, ulimits, etc etc....
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From krice at suspicious.org  Tue Dec 23 01:57:40 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:57:40 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230152u78d3ec3ci22961951e8686831@mail.gmail.com>
Message-ID: <C5761234.4B5D1%krice@suspicious.org>

Oh! Well who knows how that will affect the performance... I have never
tested it with that... Try the echo tester but be sure you are using the
media refector with sipp or you arent doing anything useful


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 10:52:09 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
> 
> the 9998 is an extension in the default.xml to test with media flowing
> through the line.
> 
> 
> 2008/12/23 Ken Rice <krice at suspicious.org>:
>> Whats this 9998 to which you refer?
>> 
>> 
>>> From: Dennis <odermann at googlemail.com>
>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>> Date: Tue, 23 Dec 2008 10:43:30 +0100
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
>>> 
>>> because the latest result was with the 9998, it can't be out app (at
>>> the moment).
>>> 
>>> so there are no other typical things or settings i could look for?
>>> 
>>> 
>>> 2008/12/23 Ken Rice <krice at suspicious.org>:
>>>> There are a number of issues you can be running into... It really depends
>>>> on
>>>> how your app works, what your actual configuration of freeswitch is, disk
>>>> IO
>>>> subsystem, ulimits, etc etc....
>>> 
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From kawarod at laposte.net  Tue Dec 23 02:02:41 2008
From: kawarod at laposte.net (rod)
Date: Tue, 23 Dec 2008 14:02:41 +0400
Subject: [Freeswitch-users] SIP Headers and use of "" in the header
In-Reply-To: <DE122F9B-6648-4549-96FC-037BFF81BB98@freeswitch.org>
References: <494FB857.9090003@laposte.net>
	<DE122F9B-6648-4549-96FC-037BFF81BB98@freeswitch.org>
Message-ID: <4950B741.8070109@laposte.net>

Thanks guys, it works.

Brian West wrote:
> <action application="export"><![CDATA[sip_h_Diversion=<123456789 at 10.10.10.254 
>  >;reason="unconditional"]]></action>
>
> /b
>
> On Dec 22, 2008, at 9:55 AM, rod wrote:
>
>   
>> Dear All,
>>
>> I've been playing with the freeswitch options for one month now, and
>> I've been able to use it with kamailio working as a registrar.
>>
>> What I'd like to do is to add a diversion header using the following
>> action in the dialplan:
>>    <action application="export"
>> data 
>> ="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="unconditional""/>
>>
>> Please note, that I'd like to put the word "unconditional" between
>> quotes, this is to comply with the SIP gateway to which I'm sending  
>> trafic.
>>
>> But I've been unable to set an escape character to use theses quotes,
>> cause as you may understand, without escape character FS will consider
>> this instruction instead:
>>
>> <action application="export"
>> data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="
>>
>> cause the quote after reason= is considered as a closing quote for  
>> data=".
>>
>> Is there a way to achieve this.
>>
>> Thanks.
>> rod
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   


From odermann at googlemail.com  Tue Dec 23 02:06:06 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 11:06:06 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C5761234.4B5D1%krice@suspicious.org>
References: <5e414ed0812230152u78d3ec3ci22961951e8686831@mail.gmail.com>
	<C5761234.4B5D1%krice@suspicious.org>
Message-ID: <5e414ed0812230206u68b6d2a1jda3c580a6692d6d4@mail.gmail.com>

sorry, i do not really understand what you mean with: "Try the echo
tester but be sure you are using the media refector with sipp or you
arent doing anything useful".

what is the "echo tester" and what is "media refector" and how could i use it?

i would like to find out, how many people can talk to each other over
the fs server at the same time. a test setup, which simulates real
calls would be very helpful for me. and then i would like to be able
to compare the results with others, to see, if everything is working
as it should.



2008/12/23 Ken Rice <krice at suspicious.org>:
> Oh! Well who knows how that will affect the performance... I have never
> tested it with that... Try the echo tester but be sure you are using the
> media refector with sipp or you arent doing anything useful


From kristjan.ugrin at gmail.com  Tue Dec 23 02:08:07 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Tue, 23 Dec 2008 11:08:07 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
	<191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
Message-ID: <op.uml03t1gatcbto@linux.site>

Today I did some more testing and packet sniffing.
When calling from google talk to google talk, packets are traveling only inside lan, there are some
queries which goes outside, but nothing more.

When using Gtalk to call someone on sip, then those packets are sent outside and I never see them again.
I think this is freeswitch configuration problem (routing?).

Where can I look further to investigate why this happens?

Thanks.


On Mon, 22 Dec 2008 18:30:46 +0100, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> if you see them leave FS and never reach dest.
> It implies a firewall somewhere in between is blocking them.
>
>
> On Mon, Dec 22, 2008 at 10:19 AM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> But what I would like to achieve is something different (quite similar).
>> You type in a message like "call 1001 at 10.99.8.20" and you it would call  
>> a
>> SIP buddy with any local number.
>>
>> In component mode you need to add a buddy everytime for a different sip
>> nr.?
>> Which would mean a lot of numbers if you would like to call more than  
>> one
>> sip nr. in a lan.
>>
>> As for the first issue, there are RTP packets traveling on FS, but never
>> reach destination after they leave our internal network.
>> Do they have to go outside lan even when the call is placed in a lan
>> between gtalk and SIP?
>> Gtalk to gtalk is no problem on same machines...
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>




From krice at suspicious.org  Tue Dec 23 02:09:32 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 04:09:32 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230206u68b6d2a1jda3c580a6692d6d4@mail.gmail.com>
Message-ID: <C57614FC.4B5D5%krice@suspicious.org>

The echo tester was refering to the echo app in freeswitch

The media reflector is part of sipp that just echos media back to the
source... That's the proper way to test media handling capabilities
otherwise you are only seeing 1/2 the media stream


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 11:06:06 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
> 
> sorry, i do not really understand what you mean with: "Try the echo
> tester but be sure you are using the media refector with sipp or you
> arent doing anything useful".
> 
> what is the "echo tester" and what is "media refector" and how could i use it?
> 
> i would like to find out, how many people can talk to each other over
> the fs server at the same time. a test setup, which simulates real
> calls would be very helpful for me. and then i would like to be able
> to compare the results with others, to see, if everything is working
> as it should.
> 
> 
> 
> 2008/12/23 Ken Rice <krice at suspicious.org>:
>> Oh! Well who knows how that will affect the performance... I have never
>> tested it with that... Try the echo tester but be sure you are using the
>> media refector with sipp or you arent doing anything useful
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From odermann at googlemail.com  Tue Dec 23 02:15:01 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 11:15:01 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C57614FC.4B5D5%krice@suspicious.org>
References: <5e414ed0812230206u68b6d2a1jda3c580a6692d6d4@mail.gmail.com>
	<C57614FC.4B5D5%krice@suspicious.org>
Message-ID: <5e414ed0812230215i5da30718md3c60c577042a8d@mail.gmail.com>

ah, that sounds interesting. so the echo app is the 9996, right?

how can i start/use the media reflector? is it something, i have to
call sipp with? sorry for this question, but i am very new in this
business.

right now i call sipp with: sipp -sn uac xx.xx.xx.xx -s 123456 -r 50
-l 400 -d 4000 -rtp_echo




2008/12/23 Ken Rice <krice at suspicious.org>:
> The echo tester was refering to the echo app in freeswitch
>
> The media reflector is part of sipp that just echos media back to the
> source... That's the proper way to test media handling capabilities
> otherwise you are only seeing 1/2 the media stream
>
>
>> From: Dennis <odermann at googlemail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 23 Dec 2008 11:06:06 +0100
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
>>
>> sorry, i do not really understand what you mean with: "Try the echo
>> tester but be sure you are using the media refector with sipp or you
>> arent doing anything useful".
>>
>> what is the "echo tester" and what is "media refector" and how could i use it?
>>
>> i would like to find out, how many people can talk to each other over
>> the fs server at the same time. a test setup, which simulates real
>> calls would be very helpful for me. and then i would like to be able
>> to compare the results with others, to see, if everything is working
>> as it should.
>>
>>
>>
>> 2008/12/23 Ken Rice <krice at suspicious.org>:
>>> Oh! Well who knows how that will affect the performance... I have never
>>> tested it with that... Try the echo tester but be sure you are using the
>>> media refector with sipp or you arent doing anything useful
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From juanbackson at gmail.com  Tue Dec 23 02:39:26 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Tue, 23 Dec 2008 18:39:26 +0800
Subject: [Freeswitch-users] Need help with "No RTP ports available!"
Message-ID: <27c25bc40812230239x7799a6a7l40e41be26a955da7@mail.gmail.com>

Hi,

I am running some stress testings on freeswitch.  When the number of
RTP ports reached around 1248 - 1250, freeswitch starts to pop out "No
RTP ports available!" error:

2008-12-23 13:14:02 [CRIT] sofia_glue.c:562
sofia_glue_tech_choose_port() No RTP ports available!

OS is Centos 5.2 64 bits and freeswitch is compiled with ./configure
--64bit options .  I also followed the wiki to maximize all my ulimit
parameters, but nothing works.  Does anyone know why?  Any help will
be greatly appreciated.

Here are my sys parameters:

[root at localhost bin]# vmstat
procs -----------memory---------- ---swap-- -----io---- --system--
-----cpu------
 r  b   swpd   free   buff  cache   si   so    bi    bo   in   cs us sy id wa st
 0  0      0 2348816 164396 1116604    0    0    14   205 2197  705  3
 3 93  1  0
[root at localhost bin]# free
             total       used       free     shared    buffers     cached
Mem:       3965952    1616748    2349204          0     164396    1116628
-/+ buffers/cache:     335724    3630228
Swap:      2031608          0    2031608
[root at localhost bin]# cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 0
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 1
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 1
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 2
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 2
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 3
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 3
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 4
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 4
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 5
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 5
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 6
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 6
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 7
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 7
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

[root at localhost bin]# cat /proc/sys/fs/file-n
cat: /proc/sys/fs/file-n: No such file or directory
[root at localhost bin]# cat /proc/sys/fs/file-
file-max  file-nr
[root at localhost bin]# cat /proc/sys/fs/file-nr
1530    0       372645
[root at localhost bin]# ulimit -a
core file size          (blocks, -c) unlimited
data seg size           (kbytes, -d) unlimited
scheduling priority             (-e) 0
file size               (blocks, -f) unlimited
pending signals                 (-i) unlimited
max locked memory       (kbytes, -l) unlimited
max memory size         (kbytes, -m) unlimited
open files                      (-n) 999999
pipe size            (512 bytes, -p) 8
POSIX message queues     (bytes, -q) unlimited
real-time priority              (-r) 0
stack size              (kbytes, -s) 244
cpu time               (seconds, -t) unlimited
max user processes              (-u) unlimited
virtual memory          (kbytes, -v) unlimited
file locks                      (-x) unlimited
[root at localhost bin]#


From alex at sinapticode.ro  Tue Dec 23 04:25:35 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Tue, 23 Dec 2008 14:25:35 +0200
Subject: [Freeswitch-users] Originate retry problem
Message-ID: <1230035135.4982.25.camel@gathern.lan>

Hi,

When I make a unsuccesfull call using session.originate, I'd like to
have a 10 minutes pause and then try again.

For our dialer we are using JS scripts, and setTimeout is not defined,
session.execute("sleep",...) doesn't work because the session has to be
originated first. And I don't really know what originate_retry_sleep_ms
does.

Basically I want a retry as described here, but with a delay between
calls: http://wiki.freeswitch.org/wiki/Busy_Call_Retry




From kristjan.ugrin at gmail.com  Tue Dec 23 06:09:34 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Tue, 23 Dec 2008 15:09:34 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.uml03t1gatcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
	<191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
	<op.uml03t1gatcbto@linux.site>
Message-ID: <op.ummb98u6atcbto@linux.site>

I've decided to do this properly:
clean fresweetch reinstall.

My worsktation hosts freeswitch + 1 sip phone also running as 1000 (linux - IP 10.99.8.221)
Other windows machine has gtalk with and also a sip phone registered as 1001 (IP 10.99.8.111).

First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works, audio is perfect.
Packets are propery travelling between 10.99.8.221 and 10.99.8.111

Second case :
On windows machine I open gtalk and I open a chat to buddy which is actually a bot logged in on freeswitch (dingaling client mode).
The I started java socket program which listens to icoming messages, after typing into client
"call 1000 at 10.99.8.221" an api command is executed:
"api originate sofia/default/1000 at 10.99.8.221 &bridge(dingaling/gmail.com/gtalk_mail(at)gmail.com)"

A call is placed between gtalk and sip phone 1000, it rings, but when both end answers there is no audio.
After a minute, the call ends itself.
I've attached wireshark dumps from both ends - what is strange is that packets are not trying to got at right IP,
instead they hit some other machine (213.x.x.x), which doesn't make sense.

Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this story):
http://pastebin.com/m75b10388

// I hope the attachments go trough - 17 KB.
test_gtalk_client_side - dump from win machine (gtalk client)
test_sip_client - dump from linux machine (freeswitch and sip phone client)

I hope to get resolved this mistery somehow.

Thank you for all kind answers.
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From ser at man.szczecin.pl  Tue Dec 23 06:20:17 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Tue, 23 Dec 2008 15:20:17 +0100
Subject: [Freeswitch-users] design of XML structure
Message-ID: <1230042017.9989.66.camel@worek.man.szczecin.pl>

Hi,

Wy there is so many ugly constructions in FS configuration files? For
example configuration of a gateway looks like this:

<include>
  <gateway name="asterlink.com">
  <param name="username" value="cluecon"/>
  <param name="realm" value="asterlink.com"/>
  <param name="from-user" value="cluecon"/>
...

This is not good structure design. I know that it gives extreme
flexibility for developers, but config files are for admins not software
developers. IMHO it should look like this:

<include>
  <gateway name="asterlink.com">
  <username>cluecon</username/>
  <realm>asterlink.com</realm/>
  <from-user>cluecon</from-user/>
...

Such structure + schema file would be a great help in configuration
editing (autocompletion and syntax check).

greetings,
Seweryn




From msc at freeswitch.org  Tue Dec 23 06:42:50 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 23 Dec 2008 06:42:50 -0800
Subject: [Freeswitch-users] design of XML structure
In-Reply-To: <1230042017.9989.66.camel@worek.man.szczecin.pl>
References: <1230042017.9989.66.camel@worek.man.szczecin.pl>
Message-ID: <06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>

Hehe, you just stepped on a land mine! There was A LOT of discussion  
about this. The simple fact of the matter is that there was no way to  
make everyone happy so the devs chose a layout that might be "ugly" to  
some. The key is that XML isn't really "pretty" anyway. The point of  
XML is that it needs to be machine readable AND human readable. Since  
humans are (usually) smarter than machines it was decided that machine  
readable is more important. Besides, when all of the GUIs get built  
you won't be hacking XML very much - if at all.

I promise you this: even if you think something is weird or curious  
about how FreeSWITCH works there is ALWAYS a good reason for the  
design decisions. Always. Nothing in FS was left to chance or caprice.

-MC

PS - this was probably a better topic for the -dev list. :)

Sent from my iPhone

On Dec 23, 2008, at 6:20 AM, Seweryn Niemiec <ser at man.szczecin.pl>  
wrote:

> Hi,
>
> Wy there is so many ugly constructions in FS configuration files? For
> example configuration of a gateway looks like this:
>
> <include>
>  <gateway name="asterlink.com">
>  <param name="username" value="cluecon"/>
>  <param name="realm" value="asterlink.com"/>
>  <param name="from-user" value="cluecon"/>
> ...
>
> This is not good structure design. I know that it gives extreme
> flexibility for developers, but config files are for admins not  
> software
> developers. IMHO it should look like this:
>
> <include>
>  <gateway name="asterlink.com">
>  <username>cluecon</username/>
>  <realm>asterlink.com</realm/>
>  <from-user>cluecon</from-user/>
> ...
>
> Such structure + schema file would be a great help in configuration
> editing (autocompletion and syntax check).
>
> greetings,
> Seweryn
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Tue Dec 23 07:03:42 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:03:42 -0600
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <495068B9.5070606@biznetnetworks.com>
References: <494F93EA.4080608@biznetnetworks.com>	<CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>	<4950625A.90100@biznetnetworks.com>
	<86D1DBA0-07B3-47A7-9CE4-B474166C5E06@freeswitch.org>
	<495068B9.5070606@biznetnetworks.com>
Message-ID: <69A3FDE3-1723-47E4-98D6-0153AB36E41E@freeswitch.org>

Na no need to close it... we can just attach everything to the  
existing jira.  ;)

/b

On Dec 22, 2008, at 10:27 PM, Pieter Eduard wrote:

> Hi Brian,
>
> sorry about the jira, i closed the too much inline log and gonna  
> open a new one after i have some results along with the wireshark  
> dump test.
>
> thank you,
>
> -Pieter-



From anthony.minessale at gmail.com  Tue Dec 23 07:03:38 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Dec 2008 09:03:38 -0600
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.ummb98u6atcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
	<191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
	<op.uml03t1gatcbto@linux.site> <op.ummb98u6atcbto@linux.site>
Message-ID: <191c3a030812230703g3133930djd2cc5ba7853bcdfb@mail.gmail.com>

when 2 devices talk via googles gtalk when they are both behind the same lan
you
are going to have problems.


on thing you can do is make an acl to ignore any candidates that are not
local
add this to your dingaling profile
<param name="candidate-acl" value="myacl"/>

then add myacl to acl.conf.xml that only allows your lan ip.

Turn off all the stun and ext-rtp-ip setting.

OR

use the windows machine from a box that is not on the sam lan behind the
same nat.




On Tue, Dec 23, 2008 at 8:09 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> I've decided to do this properly:
> clean fresweetch reinstall.
>
> My worsktation hosts freeswitch + 1 sip phone also running as 1000 (linux -
> IP 10.99.8.221)
> Other windows machine has gtalk with and also a sip phone registered as
> 1001 (IP 10.99.8.111).
>
> First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works,
> audio is perfect.
> Packets are propery travelling between 10.99.8.221 and 10.99.8.111
>
> Second case :
> On windows machine I open gtalk and I open a chat to buddy which is
> actually a bot logged in on freeswitch (dingaling client mode).
> The I started java socket program which listens to icoming messages, after
> typing into client
> "call 1000 at 10.99.8.221" an api command is executed:
> "api originate sofia/default/1000 at 10.99.8.221 &bridge(dingaling/
> gmail.com/gtalk_mail(at)gmail.com<http://gmail.com/gtalk_mail%28at%29gmail.com>
> )"
>
> A call is placed between gtalk and sip phone 1000, it rings, but when both
> end answers there is no audio.
> After a minute, the call ends itself.
> I've attached wireshark dumps from both ends - what is strange is that
> packets are not trying to got at right IP,
> instead they hit some other machine (213.x.x.x), which doesn't make sense.
>
> Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this
> story):
> http://pastebin.com/m75b10388
>
> // I hope the attachments go trough - 17 KB.
> test_gtalk_client_side - dump from win machine (gtalk client)
> test_sip_client - dump from linux machine (freeswitch and sip phone client)
>
> I hope to get resolved this mistery somehow.
>
> Thank you for all kind answers.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Dec 23 07:04:53 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:04:53 -0600
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <21140461.post@talk.nabble.com>
References: <21127913.post@talk.nabble.com>
	<2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
	<21140461.post@talk.nabble.com>
Message-ID: <490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org>

Well in this context the phones need to hangup... they aren't going to  
do so automatically.  So you'll need to hang up on them or they will  
need to hangup... or you can kick everyone from the conference with an  
api command.

/b

On Dec 23, 2008, at 1:03 AM, Carole O. wrote:

>
> Hello,
>
> When I do a "show channels" in the cli the channels to the speakers  
> are
> listed even if the speakers have stopped transmitting. If I call the
> speakers again freeswitch create new channels. If I do a "show  
> channels"
> again I can see the old and new ones. If I can keep doing this, each  
> time
> new channels are created while the old ones are still there.
> I have noticed after 5 minutes the channels that are not used  
> anymore close.
> I believed there was a kind of timeout to detect the channels that  
> are not
> in use.
>
> What I would like to know is if there is a way to close from these  
> channels
> the dialplan .
>
> Thanks,
> Carole



From brian at freeswitch.org  Tue Dec 23 07:06:30 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:06:30 -0600
Subject: [Freeswitch-users] Error when building freeswitch on Debian
	Etch 64bit.
In-Reply-To: <c564ccb60812230110j58eb7161rd0c6dc4660d828a5@mail.gmail.com>
References: <c564ccb60812230110j58eb7161rd0c6dc4660d828a5@mail.gmail.com>
Message-ID: <7B08D097-C40B-419F-BB29-C8E192A8FB87@freeswitch.org>

Please install c++ compiler.

/b

On Dec 23, 2008, at 3:10 AM, B Karthik wrote:

> Hi,
>
> I am getting the following error when compiling latest Freeswitch  
> with svn Revision no - 10914 on Debian etch 64bit. Freeswitch  
> version 1.0.1 is building successfully.
>
> Making all in .
> gcc -I/opt/src/freeswitch/src/include -I/opt/src/freeswitch/libs/ 
> libteletone/src -fPIC -Werror -g -ggdb -g -O2 -pthread -DLINUX=2 - 
> D_REENTRANT -D_GNU_SOURCE -I/opt/src/freeswitch/libs/apr/include -I/ 
> opt/src/freeswitch/libs/apr-util/include -I/opt/src/freeswitch/libs/ 
> stfu -I/opt/src/freeswitch/libs/sqlite -I/opt/src/freeswitch/libs/ 
> pcre -I/opt/src/freeswitch/libs/srtp/include -I/opt/src/freeswitch/ 
> libs/srtp/crypto/include -I/opt/src/freeswitch/libs/libresample/ 
> include -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -DSWITCH_HAVE_ODBC - 
> Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -Wall -std=c99 - 
> pedantic -o .libs/freeswitch freeswitch-switch.o  -lm -L/usr/local/ 
> lib ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl - 
> lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,-- 
> rpath -Wl,/opt/freeswitch/lib
> ./.libs/libfreeswitch.so: undefined reference to `operator  
> new(unsigned long)'
> ./.libs/libfreeswitch.so: undefined reference to `operator  
> delete(void*)'
> ./.libs/libfreeswitch.so: undefined reference to  
> `__gxx_personality_v0'
> ./.libs/libfreeswitch.so: undefined reference to `__cxa_pure_virtual'
> ./.libs/libfreeswitch.so: undefined reference to `vtable for  
> __cxxabiv1::__class_type_info'
> collect2: ld returned 1 exit status
> make[2]: *** [freeswitch] Error 1
>
> Thanks.
>
> B Karthik
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Dec 23 07:07:25 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:07:25 -0600
Subject: [Freeswitch-users] Need help with "No RTP ports available!"
In-Reply-To: <27c25bc40812230239x7799a6a7l40e41be26a955da7@mail.gmail.com>
References: <27c25bc40812230239x7799a6a7l40e41be26a955da7@mail.gmail.com>
Message-ID: <5D155529-6D4F-4AB9-9A35-45980DF4A34F@freeswitch.org>

How exactly are you load testing this?  Can you provide us an example?
/b

On Dec 23, 2008, at 4:39 AM, Juan Backson wrote:

> Hi,
>
> I am running some stress testings on freeswitch.  When the number of
> RTP ports reached around 1248 - 1250, freeswitch starts to pop out "No
> RTP ports available!" error:
>
> 2008-12-23 13:14:02 [CRIT] sofia_glue.c:562
> sofia_glue_tech_choose_port() No RTP ports available!
>
> OS is Centos 5.2 64 bits and freeswitch is compiled with ./configure
> --64bit options .  I also followed the wiki to maximize all my ulimit
> parameters, but nothing works.  Does anyone know why?  Any help will
> be greatly appreciated.
>
> Here are my sys parameters:
>
> [root at localhost bin]# vmstat
> procs -----------memory---------- ---swap-- -----io---- --system--
> -----cpu------
> r  b   swpd   free   buff  cache   si   so    bi    bo   in   cs us  
> sy id wa st
> 0  0      0 2348816 164396 1116604    0    0    14   205 2197  705  3
> 3 93  1  0
> [root at localhost bin]# free
>             total       used       free     shared    buffers      
> cached
> Mem:       3965952    1616748    2349204          0     164396     
> 1116628
> -/+ buffers/cache:     335724    3630228
> Swap:      2031608          0    2031608
> [root at localhost bin]# cat /proc/cpuinfo
> processor       : 0
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 0
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 1
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 1
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 2
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 2
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 3
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 3
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 4
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 4
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 5
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 5
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 6
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 6
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 7
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 7
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> [root at localhost bin]# cat /proc/sys/fs/file-n
> cat: /proc/sys/fs/file-n: No such file or directory
> [root at localhost bin]# cat /proc/sys/fs/file-
> file-max  file-nr
> [root at localhost bin]# cat /proc/sys/fs/file-nr
> 1530    0       372645
> [root at localhost bin]# ulimit -a
> core file size          (blocks, -c) unlimited
> data seg size           (kbytes, -d) unlimited
> scheduling priority             (-e) 0
> file size               (blocks, -f) unlimited
> pending signals                 (-i) unlimited
> max locked memory       (kbytes, -l) unlimited
> max memory size         (kbytes, -m) unlimited
> open files                      (-n) 999999
> pipe size            (512 bytes, -p) 8
> POSIX message queues     (bytes, -q) unlimited
> real-time priority              (-r) 0
> stack size              (kbytes, -s) 244
> cpu time               (seconds, -t) unlimited
> max user processes              (-u) unlimited
> virtual memory          (kbytes, -v) unlimited
> file locks                      (-x) unlimited
> [root at localhost bin]#
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Dec 23 07:13:42 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:13:42 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find	user
In-Reply-To: <20081223041237.GA19348@jdc.jasonjgw.net>
References: <200812230110.mBN1A0t6004432@jwlab.FEITH.COM>
	<20081223041237.GA19348@jdc.jasonjgw.net>
Message-ID: <55A22F7C-BEF9-4238-93CE-C5881C02B7DA@freeswitch.org>

You don't have a default user in domain 192.168.14.10, in the default  
config I used this so that you can set some vars on every call with  
one call to set_user and it would set all the vars from the default  
user on the current session.  Its best to rip it out the set_user call  
if you have modified things.

/b

On Dec 22, 2008, at 10:12 PM, Jason White wrote:

> All fine so far.
>> When I place a call from the VoIP phone FreeSWITCH complains:
>>
>>  [WARNING] switch_ivr.c:1941 switch_ivr_set_user() can't find user
>>    [default at 192.168.14.10]
>>
>> where 192.168.14.10 is the number assigned to the logical interface,
>> however the call goes through / everything seems to work.
>
> Somewhere in your dial plan, the set_user application is being  
> called with
> the above user and domain as parameter.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
>
> The logs should show you which extensions are being executed in the  
> dial plan
> so you can work out how it reached this point and why it's invoking  
> set_user
> there. Basically, work through the logic of your dial plan to find  
> out why
> this is happening.
>
> I'm sure others will have more specific advice, but, basically, it  
> has to do
> with the details of how your dial plan is configured.

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From kristjan.ugrin at gmail.com  Tue Dec 23 07:15:47 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Tue, 23 Dec 2008 16:15:47 +0100
Subject: [Freeswitch-users] [SOLVED] Call between gtalk and sip - no
	audio
In-Reply-To: <191c3a030812230703g3133930djd2cc5ba7853bcdfb@mail.gmail.com>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
	<191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
	<op.uml03t1gatcbto@linux.site> <op.ummb98u6atcbto@linux.site>
	<191c3a030812230703g3133930djd2cc5ba7853bcdfb@mail.gmail.com>
Message-ID: <op.ummfclkqatcbto@linux.site>

Thanks, commenting ext-rtp fixed my issue.
In case of further problems I'll do what you suggested.

Thank you again for all help.

On Tue, 23 Dec 2008 16:03:38 +0100, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> when 2 devices talk via googles gtalk when they are both behind the same  
> lan
> you
> are going to have problems.
>
>
> on thing you can do is make an acl to ignore any candidates that are not
> local
> add this to your dingaling profile
> <param name="candidate-acl" value="myacl"/>
>
> then add myacl to acl.conf.xml that only allows your lan ip.
>
> Turn off all the stun and ext-rtp-ip setting.
>
> OR
>
> use the windows machine from a box that is not on the sam lan behind the
> same nat.
>
>
>
>
> On Tue, Dec 23, 2008 at 8:09 AM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> I've decided to do this properly:
>> clean fresweetch reinstall.
>>
>> My worsktation hosts freeswitch + 1 sip phone also running as 1000  
>> (linux -
>> IP 10.99.8.221)
>> Other windows machine has gtalk with and also a sip phone registered as
>> 1001 (IP 10.99.8.111).
>>
>> First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works,
>> audio is perfect.
>> Packets are propery travelling between 10.99.8.221 and 10.99.8.111
>>
>> Second case :
>> On windows machine I open gtalk and I open a chat to buddy which is
>> actually a bot logged in on freeswitch (dingaling client mode).
>> The I started java socket program which listens to icoming messages,  
>> after
>> typing into client
>> "call 1000 at 10.99.8.221" an api command is executed:
>> "api originate sofia/default/1000 at 10.99.8.221 &bridge(dingaling/
>> gmail.com/gtalk_mail(at)gmail.com<http://gmail.com/gtalk_mail%28at%29gmail.com>
>> )"
>>
>> A call is placed between gtalk and sip phone 1000, it rings, but when  
>> both
>> end answers there is no audio.
>> After a minute, the call ends itself.
>> I've attached wireshark dumps from both ends - what is strange is that
>> packets are not trying to got at right IP,
>> instead they hit some other machine (213.x.x.x), which doesn't make  
>> sense.
>>
>> Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this
>> story):
>> http://pastebin.com/m75b10388
>>
>> // I hope the attachments go trough - 17 KB.
>> test_gtalk_client_side - dump from win machine (gtalk client)
>> test_sip_client - dump from linux machine (freeswitch and sip phone  
>> client)
>>
>> I hope to get resolved this mistery somehow.
>>
>> Thank you for all kind answers.
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>



-- 
Porn - the reason you need a new hard drive.




From ser at man.szczecin.pl  Tue Dec 23 07:19:36 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Tue, 23 Dec 2008 16:19:36 +0100
Subject: [Freeswitch-users] design of XML structure
In-Reply-To: <06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
References: <1230042017.9989.66.camel@worek.man.szczecin.pl>
	<06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
Message-ID: <1230045576.9989.78.camel@worek.man.szczecin.pl>

On Tue, 2008-12-23 at 06:42 -0800, Michael S Collins wrote:
> Hehe, you just stepped on a land mine! There was A LOT of discussion  

ok. I promise that this my last post on this subject :)

> about this. The simple fact of the matter is that there was no way to  
> make everyone happy so the devs chose a layout that might be "ugly" to  
> some. The key is that XML isn't really "pretty" anyway. The point of  
> XML is that it needs to be machine readable AND human readable. Since  
> humans are (usually) smarter than machines it was decided that machine  
> readable is more important. 

I like XML actually. IMHO xml with good editor and schema... there is 
nothing better for config writer.

-- 
greetings,
Seweryn



From msc at freeswitch.org  Tue Dec 23 07:52:16 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 23 Dec 2008 07:52:16 -0800
Subject: [Freeswitch-users] Originate retry problem
In-Reply-To: <1230035135.4982.25.camel@gathern.lan>
References: <1230035135.4982.25.camel@gathern.lan>
Message-ID: <AF39A731-D5D0-4543-8B95-BB271A660136@freeswitch.org>

Have you checked out 'sched_api'?

-MC

Sent from my iPhone

On Dec 23, 2008, at 4:25 AM, Alexandru Nedelcu <alex at sinapticode.ro>  
wrote:

> Hi,
>
> When I make a unsuccesfull call using session.originate, I'd like to
> have a 10 minutes pause and then try again.
>
> For our dialer we are using JS scripts, and setTimeout is not defined,
> session.execute("sleep",...) doesn't work because the session has to  
> be
> originated first. And I don't really know what  
> originate_retry_sleep_ms
> does.
>
> Basically I want a retry as described here, but with a delay between
> calls: http://wiki.freeswitch.org/wiki/Busy_Call_Retry
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Tue Dec 23 07:53:39 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 23 Dec 2008 07:53:39 -0800
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org>
References: <21127913.post@talk.nabble.com>
	<2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
	<21140461.post@talk.nabble.com>
	<490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org>
Message-ID: <97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org>

Carole,

Are you calling the hangup app from the Dialplan?

-MC

Sent from my iPhone

On Dec 23, 2008, at 7:04 AM, Brian West <brian at freeswitch.org> wrote:

> Well in this context the phones need to hangup... they aren't going to
> do so automatically.  So you'll need to hang up on them or they will
> need to hangup... or you can kick everyone from the conference with an
> api command.
>
> /b
>
> On Dec 23, 2008, at 1:03 AM, Carole O. wrote:
>
>>
>> Hello,
>>
>> When I do a "show channels" in the cli the channels to the speakers
>> are
>> listed even if the speakers have stopped transmitting. If I call the
>> speakers again freeswitch create new channels. If I do a "show
>> channels"
>> again I can see the old and new ones. If I can keep doing this, each
>> time
>> new channels are created while the old ones are still there.
>> I have noticed after 5 minutes the channels that are not used
>> anymore close.
>> I believed there was a kind of timeout to detect the channels that
>> are not
>> in use.
>>
>> What I would like to know is if there is a way to close from these
>> channels
>> the dialplan .
>>
>> Thanks,
>> Carole
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Tue Dec 23 07:57:07 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:57:07 -0600
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org>
References: <21127913.post@talk.nabble.com>
	<2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
	<21140461.post@talk.nabble.com>
	<490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org>
	<97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org>
Message-ID: <E9922E74-A4F4-47CE-B841-A1B31B526564@freeswitch.org>

I think Carole is calling a group of people into a conference..  
leaving and expecting everyone to get kicked.

/b

On Dec 23, 2008, at 9:53 AM, Michael S Collins wrote:

> Carole,
>
> Are you calling the hangup app from the Dialplan?
>
> -MC
>
> Sent from my iPhone



From chris.chen2004 at gmail.com  Tue Dec 23 10:12:16 2008
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Tue, 23 Dec 2008 13:12:16 -0500
Subject: [Freeswitch-users] Latest SVN trunk r10919 failed to build on OS X
Message-ID: <507898380812231012t4369dbcal4cfa6878be2f7c41@mail.gmail.com>

Hi all, if you have the same problem as me failing to build the latest trunk
on OS X 10.5.6
please see the error message below:

making all in .
Compiling src/switch_xml.c ...
cc1: warnings being treated as errors
src/switch_xml.c: In function 'switch_xml_find_child_multi':
src/switch_xml.c:310: warning: 'value' may be used uninitialized in this
functio
n
make[2]: *** [libfreeswitch_la-switch_xml.lo] Error 1
Making all in src
Making all in mod

making all mod_cdr_csv
make[5]: *** No rule to make target
`/Users/yunzhangchen/freeswitch/libfreeswitc
h.la', needed by `mod_cdr_csv.so'.  Stop.
make[4]: *** [all] Error 1
make[3]: *** [mod_cdr_csv-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2


Thanks

Chris
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From brian at freeswitch.org  Tue Dec 23 10:17:57 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 12:17:57 -0600
Subject: [Freeswitch-users] Latest SVN trunk r10919 failed to build on
	OS X
In-Reply-To: <507898380812231012t4369dbcal4cfa6878be2f7c41@mail.gmail.com>
References: <507898380812231012t4369dbcal4cfa6878be2f7c41@mail.gmail.com>
Message-ID: <0E399C73-2FFC-4769-A4E2-05B3F75F7ED3@freeswitch.org>

Update and try again.

/b

On Dec 23, 2008, at 12:12 PM, Chris Chen wrote:

> Hi all, if you have the same problem as me failing to build the  
> latest trunk on OS X 10.5.6
> please see the error message below:
>
> making all in .
> Compiling src/switch_xml.c ...
> cc1: warnings being treated as errors
> src/switch_xml.c: In function 'switch_xml_find_child_multi':
> src/switch_xml.c:310: warning: 'value' may be used uninitialized in  
> this functio
> n
> make[2]: *** [libfreeswitch_la-switch_xml.lo] Error 1
> Making all in src
> Making all in mod
>
> making all mod_cdr_csv
> make[5]: *** No rule to make target `/Users/yunzhangchen/freeswitch/ 
> libfreeswitc
> h.la', needed by `mod_cdr_csv.so'.  Stop.
> make[4]: *** [all] Error 1
> make[3]: *** [mod_cdr_csv-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
>  +-------- FreeSWITCH Build Complete -----------+
>  + FreeSWITCH has been successfully built.      +
>  + Install by running:                          +
>  +                                              +
>  +               make install                   +
>  +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
>
> Thanks
>
> Chris
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Tue Dec 23 14:31:15 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 24 Dec 2008 09:31:15 +1100
Subject: [Freeswitch-users] design of XML structure
In-Reply-To: <06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
References: <1230042017.9989.66.camel@worek.man.szczecin.pl>
	<06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
Message-ID: <20081223223115.GA6083@jdc.jasonjgw.net>

On Tue, Dec 23, 2008 at 06:42:50AM -0800, Michael S Collins wrote:
> Besides, when all of the GUIs get built  
> you won't be hacking XML very much - if at all.

And the XML will still be there for those of us who prefer editing
configuration files to using GUIs. I'm very much a Unix shell type of person.



From john at feith.com  Tue Dec 23 14:51:38 2008
From: john at feith.com (John Wehle)
Date: Tue, 23 Dec 2008 17:51:38 -0500 (EST)
Subject: [Freeswitch-users]  another switch_ivr_set_user() can't find user
Message-ID: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>

> You don't have a default user in domain 192.168.14.10, in the default  
> config I used this so that you can set some vars on every call with

Thanks for pointing it out and explaining the purpose.

It looks like the domain is coming from set_domain in default.xml
which gets it from sip_auth_realm.  I guess the question is if
force-register-domain is being used then:

  a) Should sip_auth_realm be set by FreeSWITCH to the value associated
     with force-register-domain

  b) or should set_domain in default.xml simply check for force-register-domain
     when setting domain?

-- John
-------------------------------------------------------------------------
|   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
|    John Wehle    |     Fax: 1-215-540-5495  |                         |
-------------------------------------------------------------------------



From brian at freeswitch.org  Tue Dec 23 15:22:27 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 17:22:27 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find user
In-Reply-To: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
References: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
Message-ID: <31BD69DD-90C5-4D7D-8826-7EE98B335F42@freeswitch.org>

You have to remember the default assumes a lot.  You go to changing  
things you have to then change the way things are assumed.

/b

On Dec 23, 2008, at 4:51 PM, John Wehle wrote:

>> You don't have a default user in domain 192.168.14.10, in the default
>> config I used this so that you can set some vars on every call with
>
> Thanks for pointing it out and explaining the purpose.
>
> It looks like the domain is coming from set_domain in default.xml
> which gets it from sip_auth_realm.  I guess the question is if
> force-register-domain is being used then:
>
>  a) Should sip_auth_realm be set by FreeSWITCH to the value associated
>     with force-register-domain
>
>  b) or should set_domain in default.xml simply check for force- 
> register-domain
>     when setting domain?
>
> -- John
> -------------------------------------------------------------------------
> |   Feith Systems  |   Voice: 1-215-646-8000  |  Email:  
> john at feith.com  |
> |    John Wehle    |     Fax: 1-215-540-5495   
> |                         |
> -------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason at jasonjgw.net  Tue Dec 23 15:27:31 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 24 Dec 2008 10:27:31 +1100
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find	user
In-Reply-To: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
References: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
Message-ID: <20081223232731.GA8541@jdc.jasonjgw.net>

On Tue, Dec 23, 2008 at 05:51:38PM -0500, John Wehle wrote:
 
> It looks like the domain is coming from set_domain in default.xml
> which gets it from sip_auth_realm.  I guess the question is if
> force-register-domain is being used then:
> 
>   a) Should sip_auth_realm be set by FreeSWITCH to the value associated
>      with force-register-domain
> 
>   b) or should set_domain in default.xml simply check for force-register-domain

As I understand it, that's a choice you need to make in designing your dial
plan.

I'm currently confronted with the same issue and, being new to FreeSWITCH, I'm
not sure what would be the best solution.

In my case, there is a SIP phone that needs to contact FreeSWITCH by its IPv4
address - there's no A record for the FreeSWITCH box, and the address is
192.168.0.2 on the local LAN. The FreeSWITCH machine does have an AAAA record
in DNS, so any incoming IPv6 call will address it by its fully-qualified
domain name. Not surprisingly, the logic in the supplied external.xml dialplan
configuration results in IPv6 incoming calls to local extensions failing, due
to the wrong domain name, after the transfer to default.xml when the bridge
application is executed. By default, of course, the domain is the IPv4
address, which is fine for IPv4 calls.

There are ways around this, of course, but I haven't worked out what would be
the best to implement.



From brian at freeswitch.org  Tue Dec 23 15:32:13 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 17:32:13 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find	user
In-Reply-To: <20081223232731.GA8541@jdc.jasonjgw.net>
References: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
	<20081223232731.GA8541@jdc.jasonjgw.net>
Message-ID: <97B3F1B3-E5C0-4EFB-A32D-38E208D40EBB@freeswitch.org>

The default is just an example that tries to get you started... the  
key thing you need to remember for ALL calls coming in that aren't  
authenticated the domain_name variable needs to be set before you  
transfer into the default context.

/b

On Dec 23, 2008, at 5:27 PM, Jason White wrote:

>
> There are ways around this, of course, but I haven't worked out what  
> would be
> the best to implement.



From msc at freeswitch.org  Tue Dec 23 16:08:48 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Dec 2008 16:08:48 -0800
Subject: [Freeswitch-users] design of XML structure
In-Reply-To: <20081223223115.GA6083@jdc.jasonjgw.net>
References: <1230042017.9989.66.camel@worek.man.szczecin.pl>
	<06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
	<20081223223115.GA6083@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812231608x41455914gc060ec5d246e5375@mail.gmail.com>

yeah, vim and emacs will always be available as your GUI. ;)
-MC

On Tue, Dec 23, 2008 at 2:31 PM, Jason White <jason at jasonjgw.net> wrote:

> On Tue, Dec 23, 2008 at 06:42:50AM -0800, Michael S Collins wrote:
> > Besides, when all of the GUIs get built
> > you won't be hacking XML very much - if at all.
>
> And the XML will still be there for those of us who prefer editing
> configuration files to using GUIs. I'm very much a Unix shell type of
> person.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Dec 23 16:46:44 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Dec 2008 16:46:44 -0800
Subject: [Freeswitch-users] FS Dev Appreciation
Message-ID: <87f2f3b90812231646y33d92da6g1beb3295c197a33c@mail.gmail.com>

FYI,

If anyone would like to show their appreciation for the FS Dev team please
email me off list so we can talk about options. I think we can all agree
that Tony & Co. have worked very hard to make FreeSWITCH a success for all
of us. Let's see if we can give a little love back to the core team!

Thanks,
Michael
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From frank at impactfax.com  Tue Dec 23 18:06:45 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 23 Dec 2008 21:06:45 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
Message-ID: <453101c9656c$45ab0900$33014c0a@ws4>

Can this command be used to run a bash script?   
I wanted to do some sox processing on some recordings after the bridge
ends and thought I should use this command.  But would like to do it in
bash. 
 
Is there a better way?
 
If this is the right way, what is the syntax for calling the bash script
with some arguments?
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From jason at jasonjgw.net  Tue Dec 23 18:25:36 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 24 Dec 2008 13:25:36 +1100
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <453101c9656c$45ab0900$33014c0a@ws4>
References: <453101c9656c$45ab0900$33014c0a@ws4>
Message-ID: <20081224022536.GA6240@jdc.jasonjgw.net>

Frank @ Impact <frank at impactfax.com> wrote:
> Can this command be used to run a bash script?   

Based on information at the wiki, this should be possible; use the system
command.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system



From msc at freeswitch.org  Tue Dec 23 22:00:24 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 23 Dec 2008 22:00:24 -0800
Subject: [Freeswitch-users] api_hangup_hook and bash
Message-ID: <AEEFD410-CEB5-4322-B04D-A7CBA5633FA4@freeswitch.org>

I'm pretty sure that this is doable. Could you give us a hint as to  
what arguments you want to send? For example, do you have one or more  
channel variables you'd like to pass to the shell script?

-MC

Sent from my iPhone

On Dec 23, 2008, at 6:25 PM, Jason White <jason at jasonjgw.net> wrote:

> Frank @ Impact <frank at impactfax.com> wrote:
>> Can this command be used to run a bash script?
>
> Based on information at the wiki, this should be possible; use the  
> system
> command.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From woodydickson at gmail.com  Wed Dec 24 00:04:34 2008
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 24 Dec 2008 16:04:34 +0800
Subject: [Freeswitch-users] Lua script directory
Message-ID: <a44688a0812240004x1a73ea49ne638628ca048900d@mail.gmail.com>

Hi,

Is it possible to change the directory where freeswitch looks for .lua
scripts?
I would like to place the lua scripts in the shared drive so multiple
freeswitch can refer to it.

Thanks,
Woody
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From jaybinks at gmail.com  Wed Dec 24 00:13:15 2008
From: jaybinks at gmail.com (jay binks)
Date: Wed, 24 Dec 2008 18:13:15 +1000
Subject: [Freeswitch-users] Lua script directory
In-Reply-To: <a44688a0812240004x1a73ea49ne638628ca048900d@mail.gmail.com>
References: <a44688a0812240004x1a73ea49ne638628ca048900d@mail.gmail.com>
Message-ID: <ec2ce17e0812240013p56b6cf63oaae2a0ac7143fed9@mail.gmail.com>

use a Dynamic link...

On Wed, Dec 24, 2008 at 6:04 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> Is it possible to change the directory where freeswitch looks for .lua
> scripts?
> I would like to place the lua scripts in the shared drive so multiple
> freeswitch can refer to it.
>
> Thanks,
> Woody
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From yudha2008 at gmail.com  Wed Dec 24 03:23:19 2008
From: yudha2008 at gmail.com (Baskar)
Date: Wed, 24 Dec 2008 16:53:19 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
Message-ID: <d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>

*Hi,

    This is my JavaScript for tone detect

session1 = new Session();
session1.originate(session1, "{ignore_early_media=false}sofia/default/
39841799874 at 172.20.191.228");
session1.execute("tone_detect", "test 400,25 r +1 hangup 'normal_clearing'
1");
session1.execute("bridge", "sofia/default/39841799874 at 172.20.191.228");
session1.execute("transfer", "39841799874");

But Tone Detect does not work at all Did i work wrongly, correct me  where i
am wrong

Another question :

In api tone detect command i got api parameters like this

api tone_detect,Start Tone Detection on a channel,<uuid> <key> <tone_spec>
[<flags> <timeout> <app> <args> <hits>]

In that I want to know what is <tone_spec> ....
I did not see any details in wiki.freeswitch site can any what should be
passed in tone_spec*.

*--
Warm Regards,
N.Baskar*
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From juanbackson at gmail.com  Wed Dec 24 04:40:02 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Wed, 24 Dec 2008 20:40:02 +0800
Subject: [Freeswitch-users] strange error while running stress testing
Message-ID: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>

Hi

I am getting the following strange error while running stress test on
freeswith.  When the number of sessions reaches 3000, I get the
following error:

2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state() RTP Error!
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]

Could someone help me out?  What do those errors mean?

Thanks in advance for all your help.

JB


From msc at freeswitch.org  Wed Dec 24 09:22:36 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Dec 2008 09:22:36 -0800
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
Message-ID: <87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>

On Wed, Dec 24, 2008 at 3:23 AM, Baskar <yudha2008 at gmail.com> wrote:

> *Hi,
>
>     This is my JavaScript for tone detect
>
> session1 = new Session();
> session1.originate(session1, "{ignore_early_media=false}sofia/default/
> 39841799874 at 172.20.191.228");
> session1.execute("tone_detect", "test 400,25 r +1 hangup 'normal_clearing'
> 1");*


Is the combination of 400Hz and 25Hz the correct busy tone for your country?
Also, the +1 means "check for tone for +1 second into the future"
Most likely you want something like +30 so that the tone detect will be
active long enough to hear something!

> *
> session1.execute("bridge", "sofia/default/39841799874 at 172.20.191.228");
> session1.execute("transfer", "39841799874");
>
> But Tone Detect does not work at all Did i work wrongly, correct me  where
> i am wrong
>
> Another question :
>
> In api tone detect command i got api parameters like this
>
> api tone_detect,Start Tone Detection on a channel,<uuid> <key> <tone_spec>
> [<flags> <timeout> <app> <args> <hits>]
>
> In that I want to know what is <tone_spec> ....*


<tone_spec> is talking about the combination of frequencies to listen for.
In your example, the tone spec is "400,25"


Hope that information helped!
-MC (mercutioviz)

*
> I did not see any details in wiki.freeswitch site can any what should be
> passed in tone_spec*.
>
> *--
> Warm Regards,
> N.Baskar*
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Dec 24 10:11:35 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Dec 2008 12:11:35 -0600
Subject: [Freeswitch-users] strange error while running stress testing
In-Reply-To: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>
References: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>
Message-ID: <191c3a030812241011i694d6483u65fe666607686da2@mail.gmail.com>

dont load test against channels that must do a stun lookup.
you are lucky you get 3000 channels doing stun. that's actually an
impressive number.
try it on a lan on a profile with no stun.


On Wed, Dec 24, 2008 at 6:40 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi
>
> I am getting the following strange error while running stress test on
> freeswith.  When the number of sessions reaches 3000, I get the
> following error:
>
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state() RTP
> Error!
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
>
> Could someone help me out?  What do those errors mean?
>
> Thanks in advance for all your help.
>
> JB
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From john at feith.com  Wed Dec 24 13:31:45 2008
From: john at feith.com (John Wehle)
Date: Wed, 24 Dec 2008 16:31:45 -0500 (EST)
Subject: [Freeswitch-users]  another switch_ivr_set_user() can't find user
Message-ID: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>

>>  a) Should sip_auth_realm be set by FreeSWITCH to the value associated
>>     with force-register-domain
>
> You have to remember the default assumes a lot.  You go to changing  
> things you have to then change the way things are assumed.

I appreciate that.  Let me ask the question slightly differently.

sofia_reg_parse_auth contains the following logic:

  if (!switch_strlen_zero(profile->reg_domain)) {
          domain_name = profile->reg_domain;
  } else {
          domain_name = realm;
  }

where profile->reg_domain is set from force-register-domain.
It then calls switch_xml_locate_user using domain_name.
It looks like force-register-domain is intended to make
FreeSWITCH believe that the user is in domain specified by
force-register-domain.

Later there's:

  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
    "sip_auth_realm", realm);
  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
    "domain_name", realm);

Shouldn't the add_header for domain_name contain the value for
the actual domain used to locate the user?

And ideally shouldn't the rest of FreeSWITCH (including examples
intended to get you started) work in the same fashion for consistency
sake (i.e. when trying to locate a user reference the domain used by
sofia_reg_parse_auth to locate the user instead of blindly using
sip_auth_realm)?

My thought is if sofia_reg_parse_auth set things up properly,
then the rest of FreeSWITCH shouldn't know or even care that
force-register-domain is in use ... it should be as if the
VoIP phone had in fact registered using the domain specified
by force-register-domain.

-- John
-------------------------------------------------------------------------
|   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
|    John Wehle    |     Fax: 1-215-540-5495  |                         |
-------------------------------------------------------------------------



From kristjan.ugrin at gmail.com  Wed Dec 24 13:51:41 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Wed, 24 Dec 2008 22:51:41 +0100
Subject: [Freeswitch-users] Behind a router
Message-ID: <op.umoscdqdatcbto@linux.site>

Hello! I installed freeswitch on different configuration, but can't get registration working.
So my phone (192.168.10.2) is now behind a wireless router (192.168.10.1),
ethernet port (192.168.0.5) is attached to eth1 (192.168.0.1).

Fs is running on that machine (it has eth0 - net and eth1 - inside lan), sofia shows:
http://pastebin.com/m6f349c32


and registration fails:
http://pastebin.com/m2916e20d

How could I fix that? So when I would call 1000 at 212.235.180.41 would ring my phone at 192.168.10.2?
I've setup port forwarding on port 5060 between 192.168.10.2 and 192.168.0.1 both ways.


-- 





From brian at freeswitch.org  Wed Dec 24 14:05:26 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Dec 2008 16:05:26 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find user
In-Reply-To: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>
References: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>
Message-ID: <91CBBB52-E51A-44C5-A2AE-EADC5A39560C@freeswitch.org>


On Dec 24, 2008, at 3:31 PM, John Wehle wrote:

>>> a) Should sip_auth_realm be set by FreeSWITCH to the value  
>>> associated
>>>    with force-register-domain
>>
>> You have to remember the default assumes a lot.  You go to changing
>> things you have to then change the way things are assumed.
>
> I appreciate that.  Let me ask the question slightly differently.
>
> sofia_reg_parse_auth contains the following logic:
>
>  if (!switch_strlen_zero(profile->reg_domain)) {
>          domain_name = profile->reg_domain;
>  } else {
>          domain_name = realm;
>  }
>
> where profile->reg_domain is set from force-register-domain.
> It then calls switch_xml_locate_user using domain_name.
> It looks like force-register-domain is intended to make
> FreeSWITCH believe that the user is in domain specified by
> force-register-domain.

Yes that is exactly what that option does. see also force-register-db- 
domain

>
>
> Later there's:
>
>  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
>    "sip_auth_realm", realm);
>  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
>    "domain_name", realm);

This looks like a typo.

> And ideally shouldn't the rest of FreeSWITCH (including examples
> intended to get you started) work in the same fashion for consistency
> sake (i.e. when trying to locate a user reference the domain used by
> sofia_reg_parse_auth to locate the user instead of blindly using
> sip_auth_realm)?

I see it the examples are rather consistent consider its SIP  
centric.   Can you provide more detail?  In FreeSWITCH for the sake of  
sanity the auth_realm is the domain name...

>
>
> My thought is if sofia_reg_parse_auth set things up properly,
> then the rest of FreeSWITCH shouldn't know or even care that
> force-register-domain is in use ... it should be as if the
> VoIP phone had in fact registered using the domain specified
> by force-register-domain.

see force-register-db-domain I think that solves the problem you're  
talking about.

/b

>
>
> -- John
> -------------------------------------------------------------------------
> |   Feith Systems  |   Voice: 1-215-646-8000  |  Email:  
> john at feith.com  |
> |    John Wehle    |     Fax: 1-215-540-5495   
> |                         |
> -------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason at jasonjgw.net  Wed Dec 24 15:34:51 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 25 Dec 2008 10:34:51 +1100
Subject: [Freeswitch-users] Setting up port audio for incoming/outgoing calls
Message-ID: <20081224233451.GA5687@jdc.jasonjgw.net>

On the wiki, an example of a port audio configuration is given that involves
creating a Sip gateway on localhost. As I couldn't get this to work
(apparently due to the external profile's detection of NAT), I thought I would
try an alternative approach. I am modifying the default dial plan here. At
some point I'll probably just rewrite it anyway.

I have created a user in the directory for extension 1020. For outbound calls,
in default.xml, I have the following:

    <extension name="portaudio" continue="true">
      <condition field="source" expression="mod_portaudio">
        <action application="set_user" data="1020@$${domain}"/>
        <action application="set" data="effective_caller_id_number=1020"/>
      </condition>
    </extension>

The log shows that the set_user is executed, as is the set
effective_caller_id_number (the latter shouldn't be necessary, unless I'm
misunderstanding).

However, running show channels after making a call from the portaudio device
still shows the user name and caller id as
FreeSWITCH,0000000000

Also, when I try to call a local extension from the audio device, I get the
following in the logs, and the call is terminated. I've checked the code, and
clearly the failure to open the file is the cause of the termination. The Sip
phone on the extension rings once and then it receives the cancellation from
FreeSWITCH.

2008-12-25 10:29:59 [DEBUG] switch_ivr_originate.c:1313 switch_ivr_originate()
P
lay Ringback File [local_stream://moh]
2008-12-25 10:29:59 [ERR] mod_local_stream.c:308 local_stream_file_open()
Unknow
n source moh
2008-12-25 10:29:59 [ERR] switch_ivr_originate.c:1322 switch_ivr_originate()
Err
or Playing File
2008-12-25 10:29:59 [DEBUG] switch_core_codec.c:122
switch_core_session_set_read
_codec() Restore original codec.
2008-12-25 10:29:59 [NOTICE] switch_ivr_originate.c:1560
switch_ivr_originate()
Hangup sofia/internal/sip:1000 at 192.168.0.4:2048;line=mxyv04us
[CS_CONSUME_MEDIA]
 [NO_ANSWER]
2008-12-25 10:29:59 [DEBUG] switch_channel.c:1494
switch_channel_perform_hangup(
) Send signal sofia/internal/sip:1000 at 192.168.0.4:2048;line=mxyv04us [KILL]

Any hints would be welcome.  There is no urgency, of course, as I'm doing this
for fun and out of interest.

Happy holidays to all on the FreeSWITCH list.



From krice at suspicious.org  Wed Dec 24 21:12:07 2008
From: krice at suspicious.org (Ken Rice)
Date: Wed, 24 Dec 2008 23:12:07 -0600
Subject: [Freeswitch-users] Happy Holidays
Message-ID: <C5787247.4B6AA%krice@suspicious.org>

Merry Christmas and Chag orim same'ach

Ken




From jason at jasonjgw.net  Wed Dec 24 22:17:38 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 25 Dec 2008 17:17:38 +1100
Subject: [Freeswitch-users] Setting up port audio for
	incoming/outgoing	calls
In-Reply-To: <20081224233451.GA5687@jdc.jasonjgw.net>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
Message-ID: <20081225061738.GA15452@jdc.jasonjgw.net>

I've solved part of my problem.

local_stream_file_open() was looking for moh/48000, because I had set the
sample rate to 48 khz in my portaudio configuration. (The context was that the
music was to be used as ring-back). Not surprisingly, the lookup failed, as
did the lookup for "moh"; if it had been moh/8000 it would have succeeded.

It all makes sense now.



From zolotov at altron.ua  Thu Dec 25 06:49:40 2008
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Thu, 25 Dec 2008 16:49:40 +0200
Subject: [Freeswitch-users] How PBXs works on different platforms
Message-ID: <1230216580.5361.9.camel@opos20.altron.lan>

  Hello!

Our experience (over 1 month) of FreeSWITCH testing under SunSolaris 10
shows us such things :


1. FreeSWITCH can be built with gcc compiler, which consists at
SunSolaris 10 by default :


$ /usr/sfw/bin/gcc --version

gcc (GCC) 3.4.3 (csl-sol210-3_4-branch+sol_rpath)

Copyright (C) 2004 Free Software Foundation, Inc.

This is free software; see the source for copying conditions. There is
NO

warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR
PURPOSE.


With this compiler is possible built FreeSWITCH except mod_openzap (this
module needs ISO C99 compatible compiler), but, it'll be shown below,
there are no any reasons to make mod_openzap under Solaris 10.


2. Newlyversion of gcc can be taken from CSW ? repositories, for example
here : 


http://mirrors.usc.edu/pub/csw/unstable/i386/5.10/


There are consists all CSW ? packages, which gcc needs for install. Full
list of CSW ? packages in a world can be found here :


http://www.opencsw.org/userguide/


So we have :


$ /opt/csw/gcc4/bin/gcc --version

gcc (GCC) 4.0.2

Copyright (C) 2005 Free Software Foundation, Inc.

This is free software; see the source for copying conditions. There is
NO

warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR
PURPOSE.


>From there we take packages aclocal & automake (they are absent into Sun
Solaris 10 distributive ), if we want build FreeSWITCH ? trunk starting
from ./bootstrap.sh .


3. One more variant of FreeSWITCH installation, which described in a few
sources in the Internet
(http://www.voiceworks.pl/cypromis/category/solaris-opensolaris/) - is
installation of Sun Studio 12 (package
SunStudio12ml-solaris-x86-200709-pkg.tar.bz2 , from Sun Microsystems
official site):


/opt/SUNWspro/bin/cc


Build can be done in a command mode, redefining parameters CC, CCOPT,
LDOPT, or even in Sun Studio GUI IDE, launching it from remote host :


# ssh -nfX <sun-host> sunstudio


4. We built & installed FreeSWITCH + Zaptel + Wanpipe on 2 64-bit
SunSolaris servers


$ isainfo -k

amd64

$ isainfo -v

64-bit amd64 applications

tscp cx16 sse3 sse2 sse fxsr amd_3dnowx amd_3dnow amd_mmx mmx cmov
amd_sysc

cx8 tsc fpu

32-bit i386 applications

tscp cx16 sse3 sse2 sse fxsr amd_3dnowx amd_3dnow amd_mmx mmx cmov
amd_sysc

cx8 tsc fpu


+ on a 64-bit server SuperMicro ? SYS-5025B; proc. Intel Xeon 3210 :


# isainfo -v

64-bit amd64 applications

ssse3 cx16 mon sse3 sse2 sse fxsr mmx cmov amd_sysc cx8 tsc fpu

32-bit i386 applications

ssse3 cx16 mon sse3 sse2 sse fxsr mmx cmov sep cx8 tsc fpu # isainfo -k

amd64


- on a last one we didn't even succeed even to start Zaptel + Wanpipe,
having exactly the same configuration like on previous two.

We didn't succeed to build some of FreeSWITCH libraries with 64-bit
support, but that is a question of time and technology. So we built
32-bit version of FreeSWITCH.


5. This variant was successfully tested for SIP calls and execution of
all extensions, whitch correctly works under Linux.


6. Then we set up support for E1/T1 Sangoma A-104. We found just this :


- ftp://ftp.sangoma.com/Solaris/Beta/SVwanpipe-i386-5.10.pkg

        - ftp://ftp.sangoma.com/Solaris/Packages/SVzaptel-i386-5.10.pkg
        


7. We managed to configure and start that variant (zaptel + TDM Voice
API). Complete equivalent of it successfully worked under Linux CentOS 5
? including configuration FreeSwitch mod_openzap.


# ./wanrouter hwprobe


-------------------------------

| Wanpipe Hardware Probe Info |

-------------------------------

1. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=1 : HWEC=0 : V=33

2. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=2 : HWEC=0 : V=33

3. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=3 : HWEC=0 : V=33

4. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=4 : HWEC=0 : V=33


Card Cnt: S508= 0 S514X= 0 S518= 0 A101-2= 0 A104= 1 A300= 0 A200= 0
A108= 0


# ./wanrouter status


Devices currently active:

wanpipe1 wanpipe2 wanpipe3 wanpipe4


Device name | Protocol | Station | Status |

wanpipe4 | AFT TE1 | N/A | Connected |

wanpipe3 | AFT TE1 | N/A | Connected |

wanpipe2 | AFT TE1 | N/A | Connected |

wanpipe1 | AFT TE1 | N/A | Connecting |


I.e. E1 spans 4, 3, 2 connected with cables are perceived by Wanpipe in
a synchronisation mode that is visible also on GREEN LEDs on the card. 


8. But we do not manage to receive any reception PRI events (under
FreeSwitch) that it is possible to receive in precisely same
configuration in Linux.


9.After that we tested just transmition of raw B-channel data with our
own tests, which has been created when we wrote Zaptel drivers for
nonstandart E1 equipment. This tests pass test data sequences into
B-channel and receives them from another B-channel, which connected to
the first B-channel with crosscable; cheks up identity of the received
information and measures delay of data passing on a loop. This tests
works good under Linux (we use them for a year), but under Solaris they
shows us that Zaptel + Wanpipe doesn't receives B-data from B-chans. 

We have source code for SVzaptel-i386-5.10.pkg , developed by SunLabs


https://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk/


Studying of this package and site has shown us such things :

      * SVzaptel-i386-5.10.pkg was developed by little community
        http://www.solarisvoip.com/;
        
      * development of this package moves very slow, last updates ? 2006
        year;
        
      * package has limited functionality (unlike original Zaptel );
        
      * SVzaptel-i386-5.10.pkg was tested on a limited amount of E1/t1
        cards (only one !) - Digium Wildcard TE110P T1/PRI).
        


We doesn't have source code for SVwanpipe- i386-5.10.pkg , so we cann't
say anything about it.


Resume:

Declared, for example: http://en.wikipedia.org/wiki/FreeSWITCH and other
URLs, platform support is realy only for SIP-protocol features but :

      * this can be done without using of any PBX, for example with the
        help of SER/OpenSER or sofia library;
        
      * is grounded that into OS is present : a) C ? compiler; b) NetBSD
        compatible IP-stack (in this case QNX 6.4 and many others OS
        could be included in the list of supported OS)
        


We suppose same things under OS *BSD family.

For objectivity it is necessary to underline that the same state of
affairs takes place and in all others free PBX, realising support E1
through the Zaptel interface: Asterisk, YATE etc. 

And all declarations about ?supported planforms? is only promotional
declarations. 

Though it is tested and affirms at :


http://www.thrallingpenguin.com/articles/asterisk-solaris.htm


- that Solaris releases of PBX should have almost in 2 times the big
productivity, than for Linux ? that has it sense only for narrow case
SIP connections that hold back testers and enter the others into error.

   Any info would be appreciated,
   Evgeniy.






From rehan at supertec.com  Thu Dec 25 05:57:32 2008
From: rehan at supertec.com (Rehan Allah Wala)
Date: Thu, 25 Dec 2008 06:57:32 -0700
Subject: [Freeswitch-users] Happy Holidays
In-Reply-To: <C5787247.4B6AA%krice@suspicious.org>
References: <C5787247.4B6AA%krice@suspicious.org>
Message-ID: <49532EDC.16933.17AC8D5C@rehan.supertec.com>

Thank You for all the wishes people,

Merry Christmas and Happy Holidays to you and every one else .

and a very Happy New Year :)

Rehan

Rehan Ahmed AllahWala
Msn/Yahoo/GoogleTalk/Email: Rehan at Rehan.com
http://www.supertec.com/ - Internet Telephony Solutions
Http://www.DIDX.net - DID Number Market Place.
Don't Remember Me ? Visit http://www.Rehan.com

~~~~~~~~~~~~~~~~~~~
"First they ignore you, then they laugh at you, then they fight you, then you win." 
By Gandhi.

"Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi



From markmorreny at gmail.com  Thu Dec 25 07:41:09 2008
From: markmorreny at gmail.com (mark morreny)
Date: Thu, 25 Dec 2008 23:41:09 +0800
Subject: [Freeswitch-users] what is going on with openmrcp?
Message-ID: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>

Hi,

I checked out Unimrcp and it seems like it is down.  Does anyone know what
is happening to Unimrcp?

It seems like mod_unimrcp is down.

I am looking for a way to integrate Freeswitch to MRCP TTS server.

Is there anywhere to do it?

Thanks,
Mark
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From brian at freeswitch.org  Thu Dec 25 08:45:54 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Dec 2008 10:45:54 -0600
Subject: [Freeswitch-users] Setting up port audio for incoming/outgoing
	calls
In-Reply-To: <20081225061738.GA15452@jdc.jasonjgw.net>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
Message-ID: <EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>

Not quite... you needed to have a moh/48000 defined in localstream  
too. ;)  so when you play local_stream://moh it appends the rate to  
the end to find the exact one.  If you define a moh by itself it would  
have fallen back to that.  See local_stream.conf.xml  ;)

/b

On Dec 25, 2008, at 12:17 AM, Jason White wrote:

> I've solved part of my problem.
>
> local_stream_file_open() was looking for moh/48000, because I had  
> set the
> sample rate to 48 khz in my portaudio configuration. (The context  
> was that the
> music was to be used as ring-back). Not surprisingly, the lookup  
> failed, as
> did the lookup for "moh"; if it had been moh/8000 it would have  
> succeeded.
>
> It all makes sense now.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Dec 25 08:48:14 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Dec 2008 10:48:14 -0600
Subject: [Freeswitch-users] what is going on with openmrcp?
In-Reply-To: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>
References: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>
Message-ID: <023F64F6-B7F0-4A06-8FBB-67B2F4AE909D@freeswitch.org>

Are you confusing openmrcp which is unsupported with unimrcp which  
doesn't have a module for freeswitch yet?

/b

On Dec 25, 2008, at 9:41 AM, mark morreny wrote:

> Hi,
>
> I checked out Unimrcp and it seems like it is down.  Does anyone  
> know what is happening to Unimrcp?
>
> It seems like mod_unimrcp is down.
>
> I am looking for a way to integrate Freeswitch to MRCP TTS server.
>
> Is there anywhere to do it?
>
> Thanks,
> Mark



From andresmartinochoa at gmail.com  Wed Dec 24 23:54:01 2008
From: andresmartinochoa at gmail.com (=?ISO-8859-1?Q?Andr=E9s_Mart=EDn_-_martyn?=)
Date: Thu, 25 Dec 2008 02:54:01 -0500
Subject: [Freeswitch-users] Behind a router
In-Reply-To: <op.umoscdqdatcbto@linux.site>
References: <op.umoscdqdatcbto@linux.site>
Message-ID: <8c1b00b30812242354i49b39ed7u44646a016d817a72@mail.gmail.com>

Hello kriko, I need a question. How do you do for create 1000 account in
freeswitch .. in witch file configuration exacly ? .. I'm beginning with
freewitch. Sorry for not create a new post on forum, but i see this
oportunity for ask you it.

Regards

On Wed, Dec 24, 2008 at 4:51 PM, kriko <kristjan.ugrin at gmail.com> wrote:

> Hello! I installed freeswitch on different configuration, but can't get
> registration working.
> So my phone (192.168.10.2) is now behind a wireless router (192.168.10.1),
> ethernet port (192.168.0.5) is attached to eth1 (192.168.0.1).
>
> Fs is running on that machine (it has eth0 - net and eth1 - inside lan),
> sofia shows:
> http://pastebin.com/m6f349c32
>
>
> and registration fails:
> http://pastebin.com/m2916e20d
>
> How could I fix that? So when I would call 1000 at 212.235.180.41 would ring
> my phone at 192.168.10.2?
> I've setup port forwarding on port 5060 between 192.168.10.2 and
> 192.168.0.1 both ways.
>
>
> --
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Andr?s Mart?n Ochoa;
passport: andresmartin at linuxmail.org;
Linux Registered User #436420;
Asterisk User Number: 1000;
PBX: (57) 1 578 20 30;
Ext: 106
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From andresmartinochoa at gmail.com  Wed Dec 24 23:55:34 2008
From: andresmartinochoa at gmail.com (=?ISO-8859-1?Q?Andr=E9s_Mart=EDn_-_martyn?=)
Date: Thu, 25 Dec 2008 02:55:34 -0500
Subject: [Freeswitch-users] Happy Holidays
In-Reply-To: <C5787247.4B6AA%krice@suspicious.org>
References: <C5787247.4B6AA%krice@suspicious.org>
Message-ID: <8c1b00b30812242355h17254b67nc7f9e460d13c895@mail.gmail.com>

Thank you. The same from Colombia :D

martyn-dev

On Thu, Dec 25, 2008 at 12:12 AM, Ken Rice <krice at suspicious.org> wrote:

> Merry Christmas and Chag orim same'ach
>
> Ken
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Andr?s Mart?n Ochoa;
passport: andresmartin at linuxmail.org;
Linux Registered User #436420;
Asterisk User Number: 1000;
PBX: (57) 1 578 20 30;
Ext: 106
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From mike at jerris.com  Thu Dec 25 09:28:58 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 25 Dec 2008 12:28:58 -0500
Subject: [Freeswitch-users] what is going on with openmrcp?
In-Reply-To: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>
References: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>
Message-ID: <80CE6997-9129-4F88-8E67-8598573AA0AE@jerris.com>


On Dec 25, 2008, at 10:41 AM, mark morreny wrote:

> Hi,
>
> I checked out Unimrcp and it seems like it is down.  Does anyone  
> know what is happening to Unimrcp?
>
> It seems like mod_unimrcp is down.

There is no such thing to my knowledge.

>
> I am looking for a way to integrate Freeswitch to MRCP TTS server.
>
> Is there anywhere to do it?
>

Currently we still support the older library openmrcp with  
mod_openmrcp.  There are thoughts to move to unimrcp in the future.


From can_man at gmx.de  Thu Dec 25 12:38:11 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Thu, 25 Dec 2008 21:38:11 +0100
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
Message-ID: <20081225203811.79240@gmx.net>

Hello,

I am trying to replace some static settings with dynamic ones which are provided by a webserver. I can bridge calls that way, however I just can't get the following transfer to work. 
The transfer works when in public.xml and looks like this:

<extension name="test9">
<condition field="destination_number" expression="^(10001)$">
<action application="set" data="domain_name=$${domain}"/>
<action application="transfer" data="9999 XML default"/>
</condition>
</extension>


The xml received by FS from the webserver looks like this:

<context name="public">
<extension name="test9">
<condition field="destination_number" expression="^(10001)$">
<action application="set" data="domain_name=$${domain}"/>
<action application="transfer" data="9999 XML default"/>
</condition>
</extension>
</context>


I have also tried without context name and extension name, but I got the same result.

This is the console log output:

2008-12-25 21:27:27 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/anonymous at sipgate.de [7281a542-d2c2-11dd-80f0-3fe65955e25b]
2008-12-25 21:27:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing anonymous->10001 in context public
2008-12-25 21:27:29 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML response is in /tmp/72829470-d2c2-11dd-80f0-3fe65955e25b.tmp.xml
2008-12-25 21:27:29 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-25 21:27:29 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 3 (sofia/external/anonymous at sipgate.de) Ended
2008-12-25 21:27:29 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]


The 9999 extension in default looks like this:

    <extension name="hold_music">
      <condition field="destination_number" expression="^9999$"/>
      <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
        <action application="answer"/>
        <action application="execute_extension" data="is_secure XML features"/>
        <action application="playback" data="$${hold_music}"/>
        <anti-action application="answer"/>
        <anti-action application="playback" data="$${hold_music}"/>
      </condition>
    </extension>


Thank you very much for your help.

Phil


-- 
Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL 
f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a


From kristjan.ugrin at gmail.com  Thu Dec 25 13:07:02 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 25 Dec 2008 22:07:02 +0100
Subject: [Freeswitch-users] Behind a router
In-Reply-To: <8c1b00b30812242354i49b39ed7u44646a016d817a72@mail.gmail.com>
References: <op.umoscdqdatcbto@linux.site>
	<8c1b00b30812242354i49b39ed7u44646a016d817a72@mail.gmail.com>
Message-ID: <op.umqkxyoeatcbto@tux.tux>

It is already there as a demo. Try to look in config folder, I'm not at my  
workstation atm.

On Thu, 25 Dec 2008 08:54:01 +0100, Andr?s Mart?n - martyn  
<andresmartinochoa at gmail.com> wrote:

> Hello kriko, I need a question. How do you do for create 1000 account in
> freeswitch .. in witch file configuration exacly ? .. I'm beginning with
> freewitch. Sorry for not create a new post on forum, but i see this
> oportunity for ask you it.
>
> Regards
>
> On Wed, Dec 24, 2008 at 4:51 PM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> Hello! I installed freeswitch on different configuration, but can't get
>> registration working.
>> So my phone (192.168.10.2) is now behind a wireless router  
>> (192.168.10.1),
>> ethernet port (192.168.0.5) is attached to eth1 (192.168.0.1).
>>
>> Fs is running on that machine (it has eth0 - net and eth1 - inside lan),
>> sofia shows:
>> http://pastebin.com/m6f349c32
>>
>>
>> and registration fails:
>> http://pastebin.com/m2916e20d
>>
>> How could I fix that? So when I would call 1000 at 212.235.180.41 would  
>> ring
>> my phone at 192.168.10.2?
>> I've setup port forwarding on port 5060 between 192.168.10.2 and
>> 192.168.0.1 both ways.
>>
>>
>> --
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>



-- 
(\__/)
(='.'=)
(")_(")


From msc at freeswitch.org  Thu Dec 25 14:12:40 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 25 Dec 2008 14:12:40 -0800
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <20081225203811.79240@gmx.net>
References: <20081225203811.79240@gmx.net>
Message-ID: <574EDB32-6665-4675-A651-B8DB58FFCAE9@freeswitch.org>

Phil,
Can you do the same test with debug turned on? F8 or "console loglevel  
debug" will do the trick.

-MC

Sent from my iPhone

On Dec 25, 2008, at 12:38 PM, can_man at gmx.de wrote:

> Hello,
>
> I am trying to replace some static settings with dynamic ones which  
> are provided by a webserver. I can bridge calls that way, however I  
> just can't get the following transfer to work.
> The transfer works when in public.xml and looks like this:
>
> <extension name="test9">
> <condition field="destination_number" expression="^(10001)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="9999 XML default"/>
> </condition>
> </extension>
>
>
> The xml received by FS from the webserver looks like this:
>
> <context name="public">
> <extension name="test9">
> <condition field="destination_number" expression="^(10001)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="9999 XML default"/>
> </condition>
> </extension>
> </context>
>
>
> I have also tried without context name and extension name, but I got  
> the same result.
>
> This is the console log output:
>
> 2008-12-25 21:27:27 [NOTICE] switch_channel.c:565  
> switch_channel_set_name() New Channel sofia/external/anonymous at sipgate.de 
>  [7281a542-d2c2-11dd-80f0-3fe65955e25b]
> 2008-12-25 21:27:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()  
> Processing anonymous->10001 in context public
> 2008-12-25 21:27:29 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML  
> response is in /tmp/72829470-d2c2-11dd-80f0-3fe65955e25b.tmp.xml
> 2008-12-25 21:27:29 [NOTICE] switch_core_state_machine.c:168  
> switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de 
>  [CS_EXECUTE] [NORMAL_CLEARING]
> 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:956  
> switch_core_session_thread() Session 3 (sofia/external/anonymous at sipgate.de 
> ) Ended
> 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:958  
> switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de 
>  [CS_HANGUP]
>
>
> The 9999 extension in default looks like this:
>
>    <extension name="hold_music">
>      <condition field="destination_number" expression="^9999$"/>
>      <condition field="${sip_has_crypto}"  
> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
>        <action application="answer"/>
>        <action application="execute_extension" data="is_secure XML  
> features"/>
>        <action application="playback" data="$${hold_music}"/>
>        <anti-action application="answer"/>
>        <anti-action application="playback" data="$${hold_music}"/>
>      </condition>
>    </extension>
>
>
> Thank you very much for your help.
>
> Phil
>
>
> -- 
> Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL
> f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T456 
> 9a
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mike at jerris.com  Thu Dec 25 14:53:32 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 25 Dec 2008 17:53:32 -0500
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <20081225203811.79240@gmx.net>
References: <20081225203811.79240@gmx.net>
Message-ID: <2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>

The $$ substitutions are only done in the static XML files.  Al

On Dec 25, 2008, at 3:38 PM, can_man at gmx.de wrote:

> Hello,
>
> I am trying to replace some static settings with dynamic ones which  
> are provided by a webserver. I can bridge calls that way, however I  
> just can't get the following transfer to work.
> The transfer works when in public.xml and looks like this:
>
> <extension name="test9">
> <condition field="destination_number" expression="^(10001)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="9999 XML default"/>
> </condition>
> </extension>
>
>
> The xml received by FS from the webserver looks like this:
>
> <context name="public">
> <extension name="test9">
> <condition field="destination_number" expression="^(10001)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="9999 XML default"/>
> </condition>
> </extension>
> </context>
>
>
> I have also tried without context name and extension name, but I got  
> the same result.
>
> This is the console log output:
>
> 2008-12-25 21:27:27 [NOTICE] switch_channel.c:565  
> switch_channel_set_name() New Channel sofia/external/anonymous at sipgate.de 
>  [7281a542-d2c2-11dd-80f0-3fe65955e25b]
> 2008-12-25 21:27:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()  
> Processing anonymous->10001 in context public
> 2008-12-25 21:27:29 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML  
> response is in /tmp/72829470-d2c2-11dd-80f0-3fe65955e25b.tmp.xml
> 2008-12-25 21:27:29 [NOTICE] switch_core_state_machine.c:168  
> switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de 
>  [CS_EXECUTE] [NORMAL_CLEARING]
> 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:956  
> switch_core_session_thread() Session 3 (sofia/external/anonymous at sipgate.de 
> ) Ended
> 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:958  
> switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de 
>  [CS_HANGUP]
>
>
> The 9999 extension in default looks like this:
>
>    <extension name="hold_music">
>      <condition field="destination_number" expression="^9999$"/>
>      <condition field="${sip_has_crypto}"  
> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
>        <action application="answer"/>
>        <action application="execute_extension" data="is_secure XML  
> features"/>
>        <action application="playback" data="$${hold_music}"/>
>        <anti-action application="answer"/>
>        <anti-action application="playback" data="$${hold_music}"/>
>      </condition>
>    </extension>
>
>
> Thank you very much for your help.
>
> Phil
>
>
> -- 
> Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL
> f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T456 
> 9a
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From can_man at gmx.de  Thu Dec 25 16:01:50 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Fri, 26 Dec 2008 01:01:50 +0100
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>
References: <20081225203811.79240@gmx.net>
	<2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>
Message-ID: <20081226000150.140990@gmx.net>


Hello,

thank you for your answers. I am a step further now, it seems that just the "condition" tags as described in the wiki aren't enough. After sending the following xml I think I get stuck at the point Micheal mentioned:

> The $$ substitutions are only done in the static XML files.  Al

FS complains that: Context default not found


XML:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="RE Dial Plan For FreeSwitch">
<context name="public">
<extension name="test10000">
<condition field="destination_number" expression="^(10000)$">
<action application="set" data="domain_name=192.168.178.22"/>
<action application="transfer" data="9999 XML default"/>
</condition>
</extension>
</context>
</section>
</document>

I also tried: <action application="set" data="domain_name=$${domain}"/>

Any idea what else I could use to make sure the default context is found?


The debug looks like this:

freeswitch at voip> 2008-12-26 00:52:09 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/anonymous at sipgate.de [0acbfb9c-d2df-11dd-9b87-537be0ec7712]
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_NEW
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State NEW
2008-12-26 00:52:09 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/external/anonymous at sipgate.de entering state [received]
2008-12-26 00:52:09 [DEBUG] sofia.c:2533 sofia_handle_sip_i_state() Remote SDP:
v=0
o=root 29259 29259 IN IP4 217.10.67.141
s=session
c=IN IP4 217.10.77.24
t=0 0
m=audio 42748 RTP/AVP 8 0 3 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=direction:active
a=nortpproxy:yes

2008-12-26 00:52:09 [DEBUG] sofia_glue.c:2409 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000]/[PCMU:0:8000]
2008-12-26 00:52:09 [DEBUG] sofia_glue.c:2409 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000]/[PCMA:8:8000]
2008-12-26 00:52:09 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/external/anonymous at sipgate.de PCMA/8000 20 ms 160 samples
2008-12-26 00:52:09 [DEBUG] sofia_glue.c:2373 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101
2008-12-26 00:52:09 [DEBUG] sofia.c:2685 sofia_handle_sip_i_state() (sofia/external/anonymous at sipgate.de) State Change CS_NEW -> CS_INIT
2008-12-26 00:52:09 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_INIT
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State INIT
2008-12-26 00:52:09 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/anonymous at sipgate.de SOFIA INIT
2008-12-26 00:52:09 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/external/anonymous at sipgate.de) State Change CS_INIT -> CS_ROUTING
2008-12-26 00:52:09 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State INIT going to sleep
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_ROUTING
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State ROUTING
2008-12-26 00:52:09 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/external/anonymous at sipgate.de SOFIA ROUTING
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/external/anonymous at sipgate.de Standard ROUTING
2008-12-26 00:52:09 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing anonymous->10000 in context public
2008-12-26 00:52:11 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML response is in /tmp/0accb9ba-d2df-11dd-9b87-537be0ec7712.tmp.xml
2008-12-26 00:52:11 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [test10000] destination_number(10000) =~ /^(10000)$/
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/external/anonymous at sipgate.de) State Change CS_ROUTING -> CS_EXECUTE
2008-12-26 00:52:11 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State ROUTING going to sleep
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_EXECUTE
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE
2008-12-26 00:52:11 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/external/anonymous at sipgate.de SOFIA EXECUTE
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:137 switch_core_standard_on_execute() sofia/external/anonymous at sipgate.de Standard EXECUTE
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/external/anonymous at sipgate.de Execute set(domain_name=192.168.178.22)
2008-12-26 00:52:11 [DEBUG] mod_dptools.c:681 set_function() sofia/external/anonymous at sipgate.de SET [domain_name]=[192.168.178.22]
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/external/anonymous at sipgate.de Execute transfer(9999 XML default)
2008-12-26 00:52:11 [DEBUG] switch_ivr.c:1245 switch_ivr_session_transfer() (sofia/external/anonymous at sipgate.de) State Change CS_EXECUTE -> CS_ROUTING
2008-12-26 00:52:11 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:11 [DEBUG] switch_ivr.c:1249 switch_ivr_session_transfer() sofia/external/anonymous at sipgate.de receive message [SWITCH_MESSAGE_INDICATE_TRANSFER]
2008-12-26 00:52:11 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:11 [NOTICE] switch_ivr.c:1251 switch_ivr_session_transfer() Transfer sofia/external/anonymous at sipgate.de to XML[9999 at default]
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE going to sleep
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_ROUTING
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State ROUTING
2008-12-26 00:52:11 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/external/anonymous at sipgate.de SOFIA ROUTING
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/external/anonymous at sipgate.de Standard ROUTING
2008-12-26 00:52:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing anonymous->9999 in context default
2008-12-26 00:52:13 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML response is in /tmp/0c0cade4-d2df-11dd-9b87-537be0ec7712.tmp.xml
2008-12-26 00:52:13 [WARNING] mod_dialplan_xml.c:263 dialplan_hunt() Context default not found
2008-12-26 00:52:13 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting
2008-12-26 00:52:13 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/external/anonymous at sipgate.de [CS_ROUTING] [NO_ROUTE_DESTINATION]
2008-12-26 00:52:13 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de [KILL]
2008-12-26 00:52:13 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State ROUTING going to sleep
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_HANGUP
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP
2008-12-26 00:52:13 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous at sipgate.de hanging up, cause: NO_ROUTE_DESTINATION
2008-12-26 00:52:13 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 404
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/anonymous at sipgate.de Standard HANGUP, cause: NO_ROUTE_DESTINATION
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP going to sleep
2008-12-26 00:52:13 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 22 (sofia/external/anonymous at sipgate.de) Locked, Waiting on external entities
2008-12-26 00:52:13 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 22 (sofia/external/anonymous at sipgate.de) Ended
2008-12-26 00:52:13 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]


Thank you.

Phil

> On Dec 25, 2008, at 3:38 PM, can_man at gmx.de wrote:
> 
> > Hello,
> >
> > I am trying to replace some static settings with dynamic ones which  
> > are provided by a webserver. I can bridge calls that way, however I  
> > just can't get the following transfer to work.
> >
> >
> > The 9999 extension in default looks like this:
> >
> >    <extension name="hold_music">
> >      <condition field="destination_number" expression="^9999$"/>
> >      <condition field="${sip_has_crypto}"  
> > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
> >        <action application="answer"/>
> >        <action application="execute_extension" data="is_secure XML  
> > features"/>
> >        <action application="playback" data="$${hold_music}"/>
> >        <anti-action application="answer"/>
> >        <anti-action application="playback" data="$${hold_music}"/>
> >      </condition>
> >    </extension>
> >
> >



-- 
Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL 
f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a


From jason at jasonjgw.net  Thu Dec 25 16:12:21 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 26 Dec 2008 11:12:21 +1100
Subject: [Freeswitch-users] Setting up port audio for
	incoming/outgoing	calls
In-Reply-To: <EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
	<EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
Message-ID: <20081226001221.GA5201@jdc.jasonjgw.net>

Thanks are due to Brian for his help with this.

Now, how do I set up my configuration for outgoing calls so that, when I make
a call from the portaudio module, the caller_id_number and caller_id_name will
be stored in the database as the extension I want, rather than as 0000000000,
FreeSWITCH?

I've tried setting origination_caller_id_name, origination_caller_id_number,
caller_id_name and caller_id_number in the dial plan, but obviously it enters
the database before these changes take effect.

Thanks to FreeSWITCH developers and other community members for your patience
and helpfulness with these inquiries, and feel free to delay your replies
until after the holidays if desired.



From brian at freeswitch.org  Thu Dec 25 16:51:16 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Dec 2008 18:51:16 -0600
Subject: [Freeswitch-users] Setting up port audio for incoming/outgoing
	calls
In-Reply-To: <20081226001221.GA5201@jdc.jasonjgw.net>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
	<EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
	<20081226001221.GA5201@jdc.jasonjgw.net>
Message-ID: <16448C42-9DDA-4063-B881-895C4BCCB873@freeswitch.org>

Open up portaudi.conf.xml and look for the callerid settings.

/b

On Dec 25, 2008, at 6:12 PM, Jason White wrote:

> Now, how do I set up my configuration for outgoing calls so that,  
> when I make
> a call from the portaudio module, the caller_id_number and  
> caller_id_name will
> be stored in the database as the extension I want, rather than as  
> 0000000000,
> FreeSWITCH?



From jason at jasonjgw.net  Thu Dec 25 17:04:47 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 26 Dec 2008 12:04:47 +1100
Subject: [Freeswitch-users] Setting up port audio for
	incoming/outgoing	calls
In-Reply-To: <16448C42-9DDA-4063-B881-895C4BCCB873@freeswitch.org>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
	<EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
	<20081226001221.GA5201@jdc.jasonjgw.net>
	<16448C42-9DDA-4063-B881-895C4BCCB873@freeswitch.org>
Message-ID: <20081226010447.GA6037@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> wrote:
> Open up portaudi.conf.xml and look for the callerid settings.

How could I possibly have missed that?

Sorry!




From brian at freeswitch.org  Thu Dec 25 17:19:25 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Dec 2008 19:19:25 -0600
Subject: [Freeswitch-users] Setting up port audio for incoming/outgoing
	calls
In-Reply-To: <20081226010447.GA6037@jdc.jasonjgw.net>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
	<EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
	<20081226001221.GA5201@jdc.jasonjgw.net>
	<16448C42-9DDA-4063-B881-895C4BCCB873@freeswitch.org>
	<20081226010447.GA6037@jdc.jasonjgw.net>
Message-ID: <FF32E0F3-C1A5-4417-9B62-B46D4617889D@freeswitch.org>

:P  happens to the best of us.

/b

On Dec 25, 2008, at 7:04 PM, Jason White wrote:

> How could I possibly have missed that?
>
> Sorry!



From markmorreny at gmail.com  Thu Dec 25 19:32:39 2008
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 26 Dec 2008 11:32:39 +0800
Subject: [Freeswitch-users] Limiting port for OpenMRCP
Message-ID: <20ad6b920812251932md7bc1dfq9cedbe61070418d8@mail.gmail.com>

Hi,

I have a dev version of a MRCP-supported TTS, but it can only allow 1 port
to connect.  Does OpenMRCP has a way to limit the number of port to connect
and queue the rest of the request somehow?

Thanks for your suggestion and help.

Mark
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From simon0922 at gmail.com  Fri Dec 26 09:22:20 2008
From: simon0922 at gmail.com (Simon Leck)
Date: Fri, 26 Dec 2008 09:22:20 -0800
Subject: [Freeswitch-users] NAT Help needed
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAAZE0eGOLxxMmp0+fzQ6K2HCgAAAEAAAAA+x9JO5/VNPm/XlGkrUG1oBAAAAAA==@gmail.com>

Hi Everybody I am a newbie in this. I have managed to setup freeSwitch but I
am unable to resolve the NAT issue. Hope somebody out there can furnish me
with guidance.

 

So far I have manage to use "Use Stun server to resolve registration problem
regarding client behind NAT device but behind two layers, NAT devices still
have problem in registration.

 

Thanks Everybody for your kind assistance.

 

Simon

Email: simon0922 at gmail.com

 

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From jason at jasonjgw.net  Thu Dec 25 21:17:51 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 26 Dec 2008 16:17:51 +1100
Subject: [Freeswitch-users] NAT Help needed
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAAZE0eGOLxxMmp0+fzQ6K2HCgAAAEAAAAA+x9JO5/VNPm/XlGkrUG1oBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAAZE0eGOLxxMmp0+fzQ6K2HCgAAAEAAAAA+x9JO5/VNPm/XlGkrUG1oBAAAAAA==@gmail.com>
Message-ID: <20081226051751.GA17995@jdc.jasonjgw.net>

I'm no NAT expert, but some NAT devices can be configured to translate the
addresses in SIP messages appropriately during routing.

Linux IPTables has a SIP conntrack option (according to my Web search, it's in
kernels >= 2.6.18) that does this.

cisco IOS also supports it - I'm not sure when that feature was introduced.



From markmorreny at gmail.com  Fri Dec 26 00:49:34 2008
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 26 Dec 2008 16:49:34 +0800
Subject: [Freeswitch-users] Need help with openmrcp setup
Message-ID: <20ad6b920812260049x28d6ad7bx4b492b06fc8ecc99@mail.gmail.com>

Hi

I tried to setup mod_openmrcp according to wiki, but I am getting the
following error:

2008-12-27 00:40:23 [ERR] switch_core_speech.c:60 switch_core_speech_open()
Invalid speech module [openmrcp]!
2008-12-27 00:40:23 [ERR] switch_ivr_play_say.c:1848 switch_ivr_speak_text()
Invalid TTS module!

Here is my dialplan

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="voicemail dialplan">
<context name="public">
<extension name="test9">
<condition field="destination_number" expression="^(.*)$">
<action application="speak" data="openmrcp:openmrcp-v2|Susan|Hello World"/>
</condition>
</extension>
</context>
</section>
</document>


I checked that mod_openmrcp.so is compiled ok:

[root at localhost bin]# ls ../mod/mod_openmrcp.so -al
-rwxr-xr-x 1 root root 4981560 Dec 26 19:56 ../mod/mod_openmrcp.so
[root at localhost bin]#

Here is what I have in the openmrcp xml config:

[root at localhost freeswitch]# cat conf/mrcp_profiles/openmrcp-v2.xml
<include>
  <profile name="openmrcp-v2" description="OpenMRCPv2 Server">
    <param name="proto_version" value="2"/>

    <param name="client_ip" value="127.0.0.1"/>
    <param name="client_port" value="8062"/>

    <param name="server_ip" value="192.168.1.109"/>
    <param name="server_port" value="5060"/>

    <param name="rtp_port_min" value="11000"/>
    <param name="rtp_port_max" value="12000"/>
  </profile>
</include>

Could someone help me out?  I would greatly appreciate any help.

Thanks,
Mark
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From yudha2008 at gmail.com  Fri Dec 26 03:16:44 2008
From: yudha2008 at gmail.com (Baskar)
Date: Fri, 26 Dec 2008 16:46:44 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
Message-ID: <d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>

*Hi Michael,
*
    * I have updated all the changes what u said, But still i did not get
any tone detect in the script *

*session1 = new Session();
session1.originate(session1, "{ignore_early_media=True}sofia/default/
39841799874 at 172.20.191.228");
session1.execute("tone_detect", "test 400 r +30 hangup 'normal_clearing'
3");
session1.execute("bridge", "sofia/default/39841799874 at 172.20.191.228");
session1.execute("transfer", "39841799874");

Still i did not know what is the error in the script or some thing
else.please guide me to run this tone detect process.

I have tried through api command also i did not get any update.
I did not get any error also but the tone detect process is not working.

I have tried some combination's like this*

*api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30
transfer '1007 XML default' 3 *
*Output:*
Content-Type: api/response
Content-Length: 45
+OK Enabling tone detection 'test' '400' 'r'

*api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30
transfer '1007' 3*
*Output:*
Content-Type: api/response
Content-Length: 45
+OK Enabling tone detection 'test' '400' 'r'


*api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30
transfer '1007 at 172.20.201.67' 3*
*Output:*
Content-Type: api/response
Content-Length: 45
+OK Enabling tone detection 'test' '400' 'r'
*
api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +1 hangup
'normal_clearing' 3*
*Output:*
Content-Type: api/response
Content-Length: 45
+OK Enabling tone detection 'test' '400' 'r'

*I get output == OK but no detect of tone.*

*"Is there any modules for tone detect to be enabled" please assist me, so
that it is useful for me!*

*Thanks in Advance.*
    *
Warm Regards,
N.Baskar
*
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From juanbackson at gmail.com  Fri Dec 26 03:33:50 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Fri, 26 Dec 2008 19:33:50 +0800
Subject: [Freeswitch-users] Hard limit on RTP sessions
Message-ID: <27c25bc40812260333o4067813bmdc6a63e1af4ffd6b@mail.gmail.com>

Hi,

Is there any hard limit set on the number of RTP sessions for
Freeswitch?  I am seeing freeswitch start sending out BYE after the
number of RTP session reaches 3000.   This problem happens even when
the machine utilization is still low.

Does anyone know what is wrong?

Thanks,
JB


From juanbackson at gmail.com  Fri Dec 26 04:23:33 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Fri, 26 Dec 2008 20:23:33 +0800
Subject: [Freeswitch-users] strange error while running stress testing
In-Reply-To: <191c3a030812241011i694d6483u65fe666607686da2@mail.gmail.com>
References: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>
	<191c3a030812241011i694d6483u65fe666607686da2@mail.gmail.com>
Message-ID: <27c25bc40812260423t22da28d9y56cc7071689b1613@mail.gmail.com>

Hi

The strange thing is that I am not using stun at all.  Also, this stun
error only pops up after the number of sessions reach a certain
amount.

JB

On Thu, Dec 25, 2008 at 2:11 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> dont load test against channels that must do a stun lookup.
> you are lucky you get 3000 channels doing stun. that's actually an
> impressive number.
> try it on a lan on a profile with no stun.
>
>
> On Wed, Dec 24, 2008 at 6:40 AM, Juan Backson <juanbackson at gmail.com> wrote:
>>
>> Hi
>>
>> I am getting the following strange error while running stress test on
>> freeswith.  When the number of sessions reaches 3000, I get the
>> following error:
>>
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state() RTP
>> Error!
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>>
>> Could someone help me out?  What do those errors mean?
>>
>> Thanks in advance for all your help.
>>
>> JB
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mike at jerris.com  Fri Dec 26 05:31:26 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 08:31:26 -0500
Subject: [Freeswitch-users] Need help with openmrcp setup
In-Reply-To: <20ad6b920812260049x28d6ad7bx4b492b06fc8ecc99@mail.gmail.com>
References: <20ad6b920812260049x28d6ad7bx4b492b06fc8ecc99@mail.gmail.com>
Message-ID: <8C88DEE7-36F6-40E6-9C17-19C22AF825F1@jerris.com>

It looks like mod_openmrcp isn't loaded.


On Dec 26, 2008, at 3:49 AM, mark morreny wrote:

> Hi
>
> I tried to setup mod_openmrcp according to wiki, but I am getting  
> the following error:
>
> 2008-12-27 00:40:23 [ERR] switch_core_speech.c:60  
> switch_core_speech_open() Invalid speech module [openmrcp]!
> 2008-12-27 00:40:23 [ERR] switch_ivr_play_say.c:1848  
> switch_ivr_speak_text() Invalid TTS module!
>
> Here is my dialplan
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="voicemail dialplan">
> <context name="public">
> <extension name="test9">
> <condition field="destination_number" expression="^(.*)$">
> <action application="speak" data="openmrcp:openmrcp-v2|Susan|Hello  
> World"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
>
> I checked that mod_openmrcp.so is compiled ok:
>
> [root at localhost bin]# ls ../mod/mod_openmrcp.so -al
> -rwxr-xr-x 1 root root 4981560 Dec 26 19:56 ../mod/mod_openmrcp.so
> [root at localhost bin]#
>
> Here is what I have in the openmrcp xml config:
>
> [root at localhost freeswitch]# cat conf/mrcp_profiles/openmrcp-v2.xml
> <include>
>   <profile name="openmrcp-v2" description="OpenMRCPv2 Server">
>     <param name="proto_version" value="2"/>
>
>     <param name="client_ip" value="127.0.0.1"/>
>     <param name="client_port" value="8062"/>
>
>     <param name="server_ip" value="192.168.1.109"/>
>     <param name="server_port" value="5060"/>
>
>     <param name="rtp_port_min" value="11000"/>
>     <param name="rtp_port_max" value="12000"/>
>   </profile>
> </include>
>
> Could someone help me out?  I would greatly appreciate any help.
>
> Thanks,
> Mark
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Dec 26 05:32:54 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 08:32:54 -0500
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
Message-ID: <BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>

Your still ignoring early media.  Are you trying to detect the tone  
before or after answer?

Mike

On Dec 26, 2008, at 6:16 AM, Baskar wrote:

> Hi Michael,
>
>      I have updated all the changes what u said, But still i did not  
> get any tone detect in the script
>
> session1 = new Session();
> session1.originate(session1, "{ignore_early_media=True}sofia/default/39841799874 at 172.20.191.228 
> ");
> session1.execute("tone_detect", "test 400 r +30 hangup  
> 'normal_clearing' 3");
> session1.execute("bridge", "sofia/default/ 
> 39841799874 at 172.20.191.228");
> session1.execute("transfer", "39841799874");
>
> Still i did not know what is the error in the script or some thing  
> else.please guide me to run this tone detect process.
>
> I have tried through api command also i did not get any update.
> I did not get any error also but the tone detect process is not  
> working.
>
> I have tried some combination's like this
>
> api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30  
> transfer '1007 XML default' 3
> Output:
> Content-Type: api/response
> Content-Length: 45
> +OK Enabling tone detection 'test' '400' 'r'
>
> api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30  
> transfer '1007' 3
> Output:
> Content-Type: api/response
> Content-Length: 45
> +OK Enabling tone detection 'test' '400' 'r'
>
>
> api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30  
> transfer '1007 at 172.20.201.67' 3
> Output:
> Content-Type: api/response
> Content-Length: 45
> +OK Enabling tone detection 'test' '400' 'r'
>
> api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +1  
> hangup 'normal_clearing' 3
> Output:
> Content-Type: api/response
> Content-Length: 45
> +OK Enabling tone detection 'test' '400' 'r'
>
> I get output == OK but no detect of tone.
>
> "Is there any modules for tone detect to be enabled" please assist  
> me, so that it is useful for me!
>
> Thanks in Advance.
>
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Dec 26 05:34:56 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 08:34:56 -0500
Subject: [Freeswitch-users] strange error while running stress testing
In-Reply-To: <27c25bc40812260423t22da28d9y56cc7071689b1613@mail.gmail.com>
References: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>
	<191c3a030812241011i694d6483u65fe666607686da2@mail.gmail.com>
	<27c25bc40812260423t22da28d9y56cc7071689b1613@mail.gmail.com>
Message-ID: <53316117-BD3B-4B8D-81A7-6FBEF61B0877@jerris.com>

You are indeed using stun.  Check vars.xml if your using the default  
configs.  Also there is config params for the rtp port range,

Mike

On Dec 26, 2008, at 7:23 AM, Juan Backson wrote:

> Hi
>
> The strange thing is that I am not using stun at all.  Also, this stun
> error only pops up after the number of sessions reach a certain
> amount.
>
> JB
>
> On Thu, Dec 25, 2008 at 2:11 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> dont load test against channels that must do a stun lookup.
>> you are lucky you get 3000 channels doing stun. that's actually an
>> impressive number.
>> try it on a lan on a profile with no stun.
>>
>>
>> On Wed, Dec 24, 2008 at 6:40 AM, Juan Backson  
>> <juanbackson at gmail.com> wrote:
>>>
>>> Hi
>>>
>>> I am getting the following strange error while running stress test  
>>> on
>>> freeswith.  When the number of sessions reaches 3000, I get the
>>> following error:
>>>
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state()  
>>> RTP
>>> Error!
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>>
>>> Could someone help me out?  What do those errors mean?
>>>
>>> Thanks in advance for all your help.
>>>
>>> JB
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From markmorreny at gmail.com  Fri Dec 26 06:08:34 2008
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 26 Dec 2008 22:08:34 +0800
Subject: [Freeswitch-users] Need help with openmrcp setup
In-Reply-To: <8C88DEE7-36F6-40E6-9C17-19C22AF825F1@jerris.com>
References: <20ad6b920812260049x28d6ad7bx4b492b06fc8ecc99@mail.gmail.com>
	<8C88DEE7-36F6-40E6-9C17-19C22AF825F1@jerris.com>
Message-ID: <20ad6b920812260608h19646e8w212eca4a249db03@mail.gmail.com>

Hi,

Thanks for the hint.  I checked the log again and found out:

2008-12-27 06:01:49 [NOTICE] switch_loadable_module.c:259
switch_loadable_module_process() Adding API Function 'lua'
2008-12-27 06:01:49 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_openmrcp.so
**/usr/local/freeswitch/mod/mod_openmrcp.so: undefined symbol:
TLSv1_method**

What is causing this problem?

Thanks for all your help.

Mark

On Fri, Dec 26, 2008 at 9:31 PM, Michael Jerris <mike at jerris.com> wrote:

> It looks like mod_openmrcp isn't loaded.
>
> On Dec 26, 2008, at 3:49 AM, mark morreny wrote:
>
> Hi
>
> I tried to setup mod_openmrcp according to wiki, but I am getting the
> following error:
>
> 2008-12-27 00:40:23 [ERR] switch_core_speech.c:60 switch_core_speech_open()
> Invalid speech module [openmrcp]!
> 2008-12-27 00:40:23 [ERR] switch_ivr_play_say.c:1848
> switch_ivr_speak_text() Invalid TTS module!
>
> Here is my dialplan
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="voicemail dialplan">
> <context name="public">
> <extension name="test9">
> <condition field="destination_number" expression="^(.*)$">
> <action application="speak" data="openmrcp:openmrcp-v2|Susan|Hello World"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
>
> I checked that mod_openmrcp.so is compiled ok:
>
> [root at localhost bin]# ls ../mod/mod_openmrcp.so -al
> -rwxr-xr-x 1 root root 4981560 Dec 26 19:56 ../mod/mod_openmrcp.so
> [root at localhost bin]#
>
> Here is what I have in the openmrcp xml config:
>
> [root at localhost freeswitch]# cat conf/mrcp_profiles/openmrcp-v2.xml
> <include>
>   <profile name="openmrcp-v2" description="OpenMRCPv2 Server">
>     <param name="proto_version" value="2"/>
>
>     <param name="client_ip" value="127.0.0.1"/>
>     <param name="client_port" value="8062"/>
>
>     <param name="server_ip" value="192.168.1.109"/>
>     <param name="server_port" value="5060"/>
>
>     <param name="rtp_port_min" value="11000"/>
>     <param name="rtp_port_max" value="12000"/>
>   </profile>
> </include>
>
> Could someone help me out?  I would greatly appreciate any help.
>
> Thanks,
> Mark
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Dec 26 08:06:03 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Dec 2008 10:06:03 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find user
In-Reply-To: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>
References: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>
Message-ID: <191c3a030812260806m2c8739eew231f832354d3aa15@mail.gmail.com>

If anything should be changed it's to add an additional
actual_register_domain header in the cases when it's being forced but it's
not completely necessary.

Typical example is when a client is using the ip address in the domain field
and you want to force it
to point at the domain name in your registry.

The point of the param is to ignore the real domain supplied in the client
and normalize all registrations to a certian domain in your db.
if you want to call registered users with this mode of operation you also
need the force-register-db-domain
which takes it a step further and writes the forced domain into the
registration db so when you try to call user/<id>@<domain>
it will find it.




On Wed, Dec 24, 2008 at 3:31 PM, John Wehle <john at feith.com> wrote:

> >>  a) Should sip_auth_realm be set by FreeSWITCH to the value associated
> >>     with force-register-domain
> >
> > You have to remember the default assumes a lot.  You go to changing
> > things you have to then change the way things are assumed.
>
> I appreciate that.  Let me ask the question slightly differently.
>
> sofia_reg_parse_auth contains the following logic:
>
>  if (!switch_strlen_zero(profile->reg_domain)) {
>          domain_name = profile->reg_domain;
>  } else {
>          domain_name = realm;
>  }
>
> where profile->reg_domain is set from force-register-domain.
> It then calls switch_xml_locate_user using domain_name.
> It looks like force-register-domain is intended to make
> FreeSWITCH believe that the user is in domain specified by
> force-register-domain.
>
> Later there's:
>
>  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
>    "sip_auth_realm", realm);
>  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
>    "domain_name", realm);
>
> Shouldn't the add_header for domain_name contain the value for
> the actual domain used to locate the user?
>
> And ideally shouldn't the rest of FreeSWITCH (including examples
> intended to get you started) work in the same fashion for consistency
> sake (i.e. when trying to locate a user reference the domain used by
> sofia_reg_parse_auth to locate the user instead of blindly using
> sip_auth_realm)?
>
> My thought is if sofia_reg_parse_auth set things up properly,
> then the rest of FreeSWITCH shouldn't know or even care that
> force-register-domain is in use ... it should be as if the
> VoIP phone had in fact registered using the domain specified
> by force-register-domain.
>
> -- John
> -------------------------------------------------------------------------
> |   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
> |    John Wehle    |     Fax: 1-215-540-5495  |                         |
> -------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Dec 26 08:11:07 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Dec 2008 10:11:07 -0600
Subject: [Freeswitch-users] Hard limit on RTP sessions
In-Reply-To: <27c25bc40812260333o4067813bmdc6a63e1af4ffd6b@mail.gmail.com>
References: <27c25bc40812260333o4067813bmdc6a63e1af4ffd6b@mail.gmail.com>
Message-ID: <191c3a030812260811x56def967lc8a68755f62b7e54@mail.gmail.com>

I think what's wrong is that instead of listening to the explanation on the
other thread you already started on this issue, you started a new thread
trying to ask the same question a different way hoping for a different
answer.  Once you learn how to configure FreeSWITCH you will be able to stop
using the profile with stun enabled for your testing.


On Fri, Dec 26, 2008 at 5:33 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> Is there any hard limit set on the number of RTP sessions for
> Freeswitch?  I am seeing freeswitch start sending out BYE after the
> number of RTP session reaches 3000.   This problem happens even when
> the machine utilization is still low.
>
> Does anyone know what is wrong?
>
> Thanks,
> JB
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From adnan at barakatdesigns.net  Fri Dec 26 08:19:51 2008
From: adnan at barakatdesigns.net (Adnan Barakat)
Date: Fri, 26 Dec 2008 16:19:51 +0000
Subject: [Freeswitch-users] group_confirm seems to be broken
Message-ID: <49550427.70606@barakatdesigns.net>

Hi all,

I've just updated from r10000 to the latest trunk (as I needed 
mod_http), and group_confirm seems to have broken after the update. Now 
the first 2-5 seconds of the file is played very quickly at poor 
quality, then the end of the file plays fine.

Here is the relevant part of the dialplan;
<action application="set" data="timer_name=soft"/>
<action application="set" data="group_confirm_key=exec"/>
<action application="set" data="group_confirm_file=playback /somefile.al"/>


The last time this happened timer_name=soft fixed the problem, but 
doesn't seem to make any difference in this case.


Thanks

Adnan


From frank at impactfax.com  Fri Dec 26 08:30:35 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 26 Dec 2008 11:30:35 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <AEEFD410-CEB5-4322-B04D-A7CBA5633FA4@freeswitch.org>
Message-ID: <565001c96777$47a318d0$33014c0a@ws4>

All I am passing into the script is the recording file name.

I tried using the system command right after the bridge command but
before a hangup command.  Thusly,

<action application="bridge" data="${enum_auto_route}"/>
<action application="system" data="/root/procrecording.sh
$${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
<action application="hangup"/>

The problem I am seeing is that sometimes this script gets run and
sometimes it does not.  I think it has to do maybe with which end hangs
up the phone.  But I cannot seem to nail it down just yet...

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins
 
I'm pretty sure that this is doable. Could you give us a hint as to  
what arguments you want to send? For example, do you have one or more  
channel variables you'd like to pass to the shell script?

-MC




From frank at impactfax.com  Fri Dec 26 08:57:15 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 26 Dec 2008 11:57:15 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <565001c96777$47a318d0$33014c0a@ws4>
Message-ID: <567c01c9677b$01739480$33014c0a@ws4>

I have confirmed that this system call does not fire if the calling
party hangs up the phone first.  Is there a way to get the script to
fire regardless of who hangs up first?

-F

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Frank @ Impact

<action application="bridge" data="${enum_auto_route}"/>
<action application="system" data="/root/procrecording.sh
$${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
<action application="hangup"/>

The problem I am seeing is that sometimes this script gets run and
sometimes it does not.  I think it has to do maybe with which end hangs
up the phone.  But I cannot seem to nail it down just yet...




From brian at freeswitch.org  Fri Dec 26 09:09:22 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Dec 2008 11:09:22 -0600
Subject: [Freeswitch-users] group_confirm seems to be broken
In-Reply-To: <49550427.70606@barakatdesigns.net>
References: <49550427.70606@barakatdesigns.net>
Message-ID: <EE30E7F5-1E96-4280-9107-5D12B3136FD7@freeswitch.org>

Please update and try again.

Committed revision 10949.

/b

On Dec 26, 2008, at 10:19 AM, Adnan Barakat wrote:

> Hi all,
>
> I've just updated from r10000 to the latest trunk (as I needed
> mod_http), and group_confirm seems to have broken after the update.  
> Now
> the first 2-5 seconds of the file is played very quickly at poor
> quality, then the end of the file plays fine.
>
> Here is the relevant part of the dialplan;
> <action application="set" data="timer_name=soft"/>
> <action application="set" data="group_confirm_key=exec"/>
> <action application="set" data="group_confirm_file=playback / 
> somefile.al"/>
>
>
> The last time this happened timer_name=soft fixed the problem, but
> doesn't seem to make any difference in this case.
>
>
> Thanks
>
> Adnan
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From adnan at barakatdesigns.net  Fri Dec 26 09:38:03 2008
From: adnan at barakatdesigns.net (Adnan Barakat)
Date: Fri, 26 Dec 2008 17:38:03 +0000
Subject: [Freeswitch-users] group_confirm seems to be broken
In-Reply-To: <EE30E7F5-1E96-4280-9107-5D12B3136FD7@freeswitch.org>
References: <49550427.70606@barakatdesigns.net>
	<EE30E7F5-1E96-4280-9107-5D12B3136FD7@freeswitch.org>
Message-ID: <4955167B.7070107@barakatdesigns.net>

Brian West wrote:
> Please update and try again.
> 
> Committed revision 10949.
Thanks Brian, works perfectly.


Adnan


From brian at freeswitch.org  Fri Dec 26 09:46:11 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Dec 2008 11:46:11 -0600
Subject: [Freeswitch-users] group_confirm seems to be broken
In-Reply-To: <4955167B.7070107@barakatdesigns.net>
References: <49550427.70606@barakatdesigns.net>
	<EE30E7F5-1E96-4280-9107-5D12B3136FD7@freeswitch.org>
	<4955167B.7070107@barakatdesigns.net>
Message-ID: <DDDB5D15-027F-44C1-99A9-B8BAF9897B50@freeswitch.org>

Next time we'll need to open a jira for tracking.  We have to get into  
that habit soon  ;)

/b

On Dec 26, 2008, at 11:38 AM, Adnan Barakat wrote:

> Brian West wrote:
>> Please update and try again.
>>
>> Committed revision 10949.
> Thanks Brian, works perfectly.
>
>
> Adnan
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From frank at impactfax.com  Fri Dec 26 11:57:18 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 26 Dec 2008 14:57:18 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <567c01c9677b$01739480$33014c0a@ws4>
Message-ID: <57ba01c96794$286de040$33014c0a@ws4>

I also tried this without success.  This will not fire at all regardless
of who hangs up.

<action application="set" data="api_hangup_hook=system /root/test.sh
$${base_dir}/recordings/9000_1.wav"/>
<action application="bridge" data="${enum_auto_route}"/>



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Frank @ Impact

 -----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Frank @ Impact

<action application="bridge" data="${enum_auto_route}"/>
<action application="system" data="/root/procrecording.sh
$${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
<action application="hangup"/>

The problem I am seeing is that sometimes this script gets run and
sometimes it does not.  I think it has to do maybe with which end hangs
up the phone.  But I cannot seem to nail it down just yet...






From msc at freeswitch.org  Fri Dec 26 12:06:26 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Dec 2008 12:06:26 -0800
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <57ba01c96794$286de040$33014c0a@ws4>
References: <567c01c9677b$01739480$33014c0a@ws4>
	<57ba01c96794$286de040$33014c0a@ws4>
Message-ID: <87f2f3b90812261206n1be8c5e3uc91df7e34961e983@mail.gmail.com>

Frank,

I'm going to check this out as soon as I can get my test system back on
line. Thanks.
-MC

P.S. - what FS version and OS version are you on? I test with latest trunk
and CentOS 5.2

On Fri, Dec 26, 2008 at 11:57 AM, Frank @ Impact <frank at impactfax.com>wrote:

> I also tried this without success.  This will not fire at all regardless
> of who hangs up.
>
> <action application="set" data="api_hangup_hook=system /root/test.sh
> $${base_dir}/recordings/9000_1.wav"/>
> <action application="bridge" data="${enum_auto_route}"/>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Frank @ Impact
>
>  -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Frank @ Impact
>
> <action application="bridge" data="${enum_auto_route}"/>
> <action application="system" data="/root/procrecording.sh
> $${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
> <action application="hangup"/>
>
> The problem I am seeing is that sometimes this script gets run and
> sometimes it does not.  I think it has to do maybe with which end hangs
> up the phone.  But I cannot seem to nail it down just yet...
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Dec 26 12:06:37 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 15:06:37 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <567c01c9677b$01739480$33014c0a@ws4>
References: <567c01c9677b$01739480$33014c0a@ws4>
Message-ID: <CB1FDFB1-27DB-4004-8EEF-CAF82BAFDC1B@jerris.com>

This is correct, if the a leg hangs up it will not continue to run the  
dialplan actions.


On Dec 26, 2008, at 11:57 AM, Frank @ Impact wrote:

> I have confirmed that this system call does not fire if the calling
> party hangs up the phone first.  Is there a way to get the script to
> fire regardless of who hangs up first?
>
> -F
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Frank @ Impact
>
> <action application="bridge" data="${enum_auto_route}"/>
> <action application="system" data="/root/procrecording.sh
> $${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
> <action application="hangup"/>
>
> The problem I am seeing is that sometimes this script gets run and
> sometimes it does not.  I think it has to do maybe with which end  
> hangs
> up the phone.  But I cannot seem to nail it down just yet...



From mike at jerris.com  Fri Dec 26 12:08:17 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 15:08:17 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <57ba01c96794$286de040$33014c0a@ws4>
References: <57ba01c96794$286de040$33014c0a@ws4>
Message-ID: <572AC073-837D-4CF3-9346-F4FA49B04E91@jerris.com>

This should work, is there any debug output at hangup that would  
indicate why it doesn't run?

Mike

On Dec 26, 2008, at 2:57 PM, Frank @ Impact wrote:

> I also tried this without success.  This will not fire at all  
> regardless
> of who hangs up.
>
> <action application="set" data="api_hangup_hook=system /root/test.sh
> $${base_dir}/recordings/9000_1.wav"/>
> <action application="bridge" data="${enum_auto_route}"/>
>



From frank at impactfax.com  Fri Dec 26 12:20:45 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 26 Dec 2008 15:20:45 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <CB1FDFB1-27DB-4004-8EEF-CAF82BAFDC1B@jerris.com>
Message-ID: <57cd01c96797$6f589a10$33014c0a@ws4>

I also tried to add this

<action application="set" data="hangup_after_bridge=false"/>

to keep the dialplan process on a-leg hangup.  But that did not work
either.

Svn 10960 is what I am testing.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
 
This is correct, if the a leg hangs up it will not continue to run the  
dialplan actions.





From mike at jerris.com  Fri Dec 26 12:40:18 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 15:40:18 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <57cd01c96797$6f589a10$33014c0a@ws4>
References: <57cd01c96797$6f589a10$33014c0a@ws4>
Message-ID: <0BB8151C-9D6B-4F5E-BADB-DC4293A6A8B4@jerris.com>

There is no way to make the dialplan continue to run when you hang up  
the a leg, that is correct.

Mike

On Dec 26, 2008, at 3:20 PM, Frank @ Impact wrote:

> I also tried to add this
>
> <action application="set" data="hangup_after_bridge=false"/>
>
> to keep the dialplan process on a-leg hangup.  But that did not work
> either.
>
> Svn 10960 is what I am testing.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
>
> This is correct, if the a leg hangs up it will not continue to run the
> dialplan actions.
>



From juanbackson at gmail.com  Sat Dec 27 00:27:43 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Sat, 27 Dec 2008 16:27:43 +0800
Subject: [Freeswitch-users] Hard limit on RTP sessions
In-Reply-To: <191c3a030812260811x56def967lc8a68755f62b7e54@mail.gmail.com>
References: <27c25bc40812260333o4067813bmdc6a63e1af4ffd6b@mail.gmail.com>
	<191c3a030812260811x56def967lc8a68755f62b7e54@mail.gmail.com>
Message-ID: <27c25bc40812270027gf6f2b8et9e1d2f9e71ac9b25@mail.gmail.com>

Hi,

Sorry for the confusion.  The 1st email was due to another problem
that I fixed, and after I fixed that problem, I am getting another
one.  Therefore, I thought the two issues are not the same.

The first issue was due to freeswich config problem related to STUN
setting.  The 2nd issue, which is what this email is about, is due to
incorrect config on sipp.  I also documented the problem and solution
in the wiki as well.

Now, I am getting another issue.  After the above two problems are
fixed, I am getting "2008-12-27 11:21:15 [ERR] switch_core_io.c:591
switch_core_session_write_frame() sofia/internal/12969 has no write
codec."  when running on a high load.

It does not happen all the time and this error does not seem to cause
call to be dropped.

Thanks for your help in advance.

JB





On Sat, Dec 27, 2008 at 12:11 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I think what's wrong is that instead of listening to the explanation on the
> other thread you already started on this issue, you started a new thread
> trying to ask the same question a different way hoping for a different
> answer.  Once you learn how to configure FreeSWITCH you will be able to stop
> using the profile with stun enabled for your testing.
>
>
> On Fri, Dec 26, 2008 at 5:33 AM, Juan Backson <juanbackson at gmail.com> wrote:
>>
>> Hi,
>>
>> Is there any hard limit set on the number of RTP sessions for
>> Freeswitch?  I am seeing freeswitch start sending out BYE after the
>> number of RTP session reaches 3000.   This problem happens even when
>> the machine utilization is still low.
>>
>> Does anyone know what is wrong?
>>
>> Thanks,
>> JB
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From yudha2008 at gmail.com  Sat Dec 27 02:56:58 2008
From: yudha2008 at gmail.com (Baskar)
Date: Sat, 27 Dec 2008 16:26:58 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
Message-ID: <d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>

*Hi **Michael,*
*
    "I try to detect the tone before answering the call.

     Is there any module for tone detect to be enabled"*

*     " I have set ignore_early_media=False **(False is case sensitive?)*"

*     But still no Tone is Detected.*

*--
Warm Regards,
N.Baskar
*
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From mike at jerris.com  Sat Dec 27 09:44:31 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 27 Dec 2008 12:44:31 -0500
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
Message-ID: <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>


On Dec 27, 2008, at 5:56 AM, Baskar wrote:

> Hi Michael,
>
>     "I try to detect the tone before answering the call.
>
>      Is there any module for tone detect to be enabled"

I believe its in mod_dptools, if its not throwing an error that the  
application does not exist, then its fine.

>
>
>      " I have set ignore_early_media=False (False is case sensitive?)"

you can omit ignore_early_media, it defaults to false.

>      But still no Tone is Detected.

Have you confirmed that tone frequency for your busy tone in your  
country and that the tone is actually being played as part of early  
media?
>
Mike

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From frank at impactfax.com  Sat Dec 27 14:21:23 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 27 Dec 2008 17:21:23 -0500
Subject: [Freeswitch-users] lua call to stop_record_session - INVALID COMMAND
Message-ID: <5e8a01c96871$73ec9f10$33014c0a@ws4>

I was trying to stop a session record from lua but when I try I get a 
"Result is INVALID COMMAND!"

I am calling this lua script with 
<action application="set" data="api_hangup_hook=lua proc.lua .."/>

so by the time the lua is called, someone has hungup one of the legs.

In the lua script I am using this to try to end the record session to
the wav file so it gets closed and so I can convert it to mp3 right away
and a few other things in the lua script...

apicmd = "stop_record_session";
apiarg = recordfile;
res = api:execute(apicmd,apiarg);

but that is when I get the INVALID COMMAND on the freeswitch console.

Is there a proper way to do this from lua?



From mike at jerris.com  Sat Dec 27 17:53:21 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 27 Dec 2008 20:53:21 -0500
Subject: [Freeswitch-users] lua call to stop_record_session - INVALID
	COMMAND
In-Reply-To: <5e8a01c96871$73ec9f10$33014c0a@ws4>
References: <5e8a01c96871$73ec9f10$33014c0a@ws4>
Message-ID: <230AF946-21EE-4D02-912F-ABD5A734C0EB@jerris.com>


On Dec 27, 2008, at 5:21 PM, Frank @ Impact wrote:

> I was trying to stop a session record from lua but when I try I get a
> "Result is INVALID COMMAND!"
>
> I am calling this lua script with
> <action application="set" data="api_hangup_hook=lua proc.lua .."/>
>
> so by the time the lua is called, someone has hungup one of the legs.
>
> In the lua script I am using this to try to end the record session to
> the wav file so it gets closed and so I can convert it to mp3 right  
> away
> and a few other things in the lua script...
>
> apicmd = "stop_record_session";
> apiarg = recordfile;
> res = api:execute(apicmd,apiarg);
>
> but that is when I get the INVALID COMMAND on the freeswitch console.
>
> Is there a proper way to do this from lua?

When the call hangs up the record session should stop by itself  
anyways so this should not be necessary.

Mike



From wiltingtree at gmail.com  Sat Dec 27 18:15:13 2008
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 27 Dec 2008 21:15:13 -0500
Subject: [Freeswitch-users] onInputCallback unstable in Python
Message-ID: <eb7797d70812271815xdf10c72v5ea1b25098205f05@mail.gmail.com>

I'm using builds 10724 and 10914 to place an outbound call from the
FreeSwitch console and use the onInputCallback functionality. My goal is to
get mod_vmd working for me.

When I run my script and press a touchtone to invoke the callback function,
I get some unstable behavior; sometimes it works fine, sometimes I get a
core dump (send me an email at wiltingtree at gmail.com if anybody wants the
core dump),  sometimes it gives me the following error:

   TypeError: onInputCallback() takes exactly 3 arguments (0 given)

In the documentation onInputCallback() takes 3 arguments, and I don't see
how it would be very useful with zero.

Here is a test script I put together which shows this behavior:

import os
from freeswitch import *
def onInputCallback(session, what, obj):
  consoleLog("INFO","IM IN THE CALLBACK!\n")
  return("continue")

def fsapi(session, stream, env, args):
  consoleLog("INFO","Hello there!!!\n")
  session =
Session("{ignore_early_media=true}sofia/gateway/gafachi/1xxxxxxxxxx")
  session.sleep(500)
  session.setInputCallback(onInputCallback)
  session.streamFile("/root/intro.wav")
  consoleLog("info","Bye!\n")
  session.hangup()
  return(session)
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From frank at impactfax.com  Sun Dec 28 08:14:27 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sun, 28 Dec 2008 11:14:27 -0500
Subject: [Freeswitch-users] lua call to stop_record_session -
	INVALIDCOMMAND
In-Reply-To: <230AF946-21EE-4D02-912F-ABD5A734C0EB@jerris.com>
Message-ID: <62c401c96907$5bad2050$33014c0a@ws4>

Mike,
I did some testing and this file is not getting closed.  I called the
script on hangup.  Made sure both legs hungup and then even did a sleep
for 5 secs to make sure FS could close any files is needed to.  Then I
made a copy of the wav file to a tmp file.  Then ended the script to
return back to the dialplan and made another copy of the wav file to a
second tmp file.

The first copy I made could not be opened by the media player. Said it
was corrupt.

The second copy of the file could be opened just fine by the media
player.

So FS is doing something to the recording file after the lua script
returns.  This is why I was trying to stop the recording session.

Any ideas?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, December 27, 2008 8:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua call to stop_record_session -
INVALIDCOMMAND

>
> apicmd = "stop_record_session";
> apiarg = recordfile;
> res = api:execute(apicmd,apiarg);
>
> but that is when I get the INVALID COMMAND on the freeswitch console.
>
> Is there a proper way to do this from lua?

When the call hangs up the record session should stop by itself  
anyways so this should not be necessary.

Mike





From frank at impactfax.com  Sun Dec 28 08:17:26 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sun, 28 Dec 2008 11:17:26 -0500
Subject: [Freeswitch-users] lua call to stop_record_session -
	INVALIDCOMMAND
In-Reply-To: <230AF946-21EE-4D02-912F-ABD5A734C0EB@jerris.com>
Message-ID: <62c501c96907$c648d800$33014c0a@ws4>

Another way I thought about doing this was to try to do a
api_hangup_hook=transfer stop_rec_exten.  Then in that extension, do the
stop recording and call my system processing script from there.

But I could not get the api hook to transfer to the designated extension
on hangup.  Is transfer not a valid call from api_hangup_hook?



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
 

On Dec 27, 2008, at 5:21 PM, Frank @ Impact wrote:

> I was trying to stop a session record from lua but when I try I get a
> "Result is INVALID COMMAND!"
>
When the call hangs up the record session should stop by itself  
anyways so this should not be necessary.

Mike




From frank at impactfax.com  Sun Dec 28 11:06:37 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sun, 28 Dec 2008 14:06:37 -0500
Subject: [Freeswitch-users] session_record post-processing
Message-ID: <636b01c9691f$68802120$33014c0a@ws4>

Maybe I am going about this all wrong.  All I am trying to do is process
a recording file of a session after either one of the legs hangs up and
the call is over.  I am just trying to convert the wav to mp3 and email
it off.  So I have a bash script to do this.  The dialplan is simple
enought
 
          <action application="set" data="RECORD_ANSWER_REQ=true"/>
          <action application="set" data="RECORD_STEREO=true"/>
          <action application="record_session"
data="$${base_dir}/recordings/3329000_1.wav"/>
          <action application="bridge" data="${enum_auto_route}"/>
          <action application="hangup"/>
 
using FS svn 10960
 
But nothing I have tried seems to get it done.  I have tried to use
api_hangup_hook to call a lua script.  But the wav file is not yet
closed for some reason yet and I cannot seem to close it in the lua
script.  Basically I get the same problem if I use 'system' with the
api_hangup_hook to call the bash script to process the recording.
 
I have tried to use the transfer application with the api_hangup_hook to
allow me to stop_record_session and then a system call from another
extension, but the transfer never happens on hangup.
 
Am I missing a simple way to do this?  Is there something similar to the
'h' extension in asterisk maybe?
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From ronmccar at gmail.com  Sun Dec 28 10:52:15 2008
From: ronmccar at gmail.com (Ron McCarthy)
Date: Sun, 28 Dec 2008 11:52:15 -0700
Subject: [Freeswitch-users] Multiple context without using directory
Message-ID: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>

Hi all,

I would like to setup FS to have many context, basically we just want to
switch calls, and since we have been using Asterisk we want to keep the
context names the same, and well it's easier it seems to me.

This is for termination only, so just sending calls out, I would think FS
can do it the same way Asterisk does, where if the IP matches then it will
use those settings with that gateway and diaplan since the context ties them
all together. No matter what I try the context always goes to default, I
even took off all the ACL's for internal and external profiles and tried
gateways in each, still no luck, i can't get around the default context. We
are going to be using IP based auth only, so no user/pass's ever.

Any help on this would be great, I have searched and searched and everyone
just uses the default context, but I am trying to avoid that as the diaplan
will get quite long as I will have to all kinds of crazy matching, when a
context is much simpler, and I would think faster then using lots of
dialplan logic.

Thanks for any help on this!

Ron
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From brian at freeswitch.org  Sun Dec 28 15:20:40 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Dec 2008 17:20:40 -0600
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
Message-ID: <44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>


On Dec 28, 2008, at 12:52 PM, Ron McCarthy wrote:

> Hi all,
>
> I would like to setup FS to have many context, basically we just  
> want to switch calls, and since we have been using Asterisk we want  
> to keep the context names the same, and well it's easier it seems to  
> me.
>
> This is for termination only, so just sending calls out, I would  
> think FS can do it the same way Asterisk does, where if the IP  
> matches then it will use those settings with that gateway and  
> diaplan since the context ties them all together. No matter what I  
> try the context always goes to default, I even took off all the  
> ACL's for internal and external profiles and tried gateways in each,  
> still no luck, i can't get around the default context. We are going  
> to be using IP based auth only, so no user/pass's ever.


This type of authentication is still tied to the directory.  If you  
notice in the default config you'll notice the domains acl... that ACL  
is built off the cidr= attribute on the user tag in the directory.

   <user id="1000" cidr="1.2.3.4/24">
     <variables>
       <variable name="user_context" value="my_context"/>
     </variables>
   </user>

So now when you "reloadacl reloadxml" yes both together like that.

I also just fixed a bug related to this due to recent directory layout  
changes.

/b

PS you can also set the context to "_domain_" and it'll auto on the  
from host.


>
>
> Any help on this would be great, I have searched and searched and  
> everyone just uses the default context, but I am trying to avoid  
> that as the diaplan will get quite long as I will have to all kinds  
> of crazy matching, when a context is much simpler, and I would think  
> faster then using lots of dialplan logic.
>
> Thanks for any help on this!
>
> Ron
> _____



From Laurent.Fabre at kirranet.com  Sun Dec 28 15:36:33 2008
From: Laurent.Fabre at kirranet.com (Laurent Fabre)
Date: Mon, 29 Dec 2008 00:36:33 +0100
Subject: [Freeswitch-users] RE :  Freeswitch/Sofia configuration problem
In-Reply-To: <87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>
References: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>,
	<87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>
Message-ID: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CDD@CARBON.kirranet.msft>

My bad, I did RTFM and everything is working great with a lot of features turned on. I have one issue with ODBC regarding FreeTDS/SQL2005 but I googled it and it seems it's a well documented one (Invalid state cursor thingie).

If anybody managed to make things work with SQL2005 and is kind enough to share tips with me, please contact me off-list. Otherwise, I'll just fallback to some other backend solution supported by .NET.

BTW, Freeswith rocks, you guys did a tremendous work!!

Happy holidays everyone.

Regards,

-- Laurent FABRE
Directeur g?n?ral
10, rue d'Aumale
75009 Paris
Tel: +33.(0)1.42.81.28.20
Mob: +33.(0)6.75.75.02.96
Fax: +33.(0)1.70.24.74.61
laurent.fabre at kirranet.com
________________________________
De : freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] de la part de Michael Collins [msc at freeswitch.org]
Date d'envoi : mardi 23 d?cembre 2008 00:22
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] Freeswitch/Sofia configuration problem

Are you using the default config? If you've made any changes at all we'd need to know about them. Also, can you turn on SIP trace so that we can see exactly what is coming and going?
Start FS like this:
TPORT_LOG=1 ./freeswitch
Press F8 to put the console in debug mode
then capture the output while you observe the bad behavior
Please put all that, plus any config changes, into a pastebin: pastebin.freeswitch.org<http://pastebin.freeswitch.org>

I'm sure there are people around here who can help you figure out what is going on.
-MC

On Mon, Dec 22, 2008 at 8:20 AM, Laurent Fabre <Laurent.Fabre at kirranet.com<mailto:Laurent.Fabre at kirranet.com>> wrote:

Hello,

I've been trying to figure out for a few days why my freeswitch instance suddenly become insensitive to SIP packets without any warning.

What usually happen is the following :

1) start just fine in foreground mode and no errors
2) wait anywhere between 2 seconds and 20 minutes
3) Sofia suddenly decide to reload everything for some reason
4) Sofia start processing SIP packets
5) work for an hour or so
6) Sofia suddenly decide to reload everything for some reason
7) become unresponsive again
8) goto 2

Both interfaces have public IP addresses assigned in a static manner (no DHCP).

I can see the SIP UDP & TCP requests comming from the phones on several sites on the wire.
The SIP TCP requests get RST in reply which is mean :(

There was a point in my setup where it would not happen but since I'm new to freeswitch I'm having an hard time backtracking.

I was wondering if iproute/tc and iptables were the culprits but I flushed everything (even rebooted without loading the rules) and it still doesn't work.

I thought some database was corrupt so I shutdown'd freeswitch and delete his db folder, no effect.

My server runs Debian 4.0etch for amd64, built freeswitch from SVN trunk.

Any pointers, help, cure against headaches would be great :)

Regards,

Laurent


_______________________________________________
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From ronmccar at gmail.com  Sun Dec 28 15:54:02 2008
From: ronmccar at gmail.com (Ron McCarthy)
Date: Sun, 28 Dec 2008 16:54:02 -0700
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
Message-ID: <3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>

I see the "brian.xml" example has it, didn't check that one, whoops.

Now I have added a user in the directory with the correct CIDR attribute,
yet when I send the call it seems to not use the directory, the calls is
coming from a Asterisk box, I have the Asterisk box pointed to the
"internal" profile's IP address, which I assume the directory would use, and
it gets rejected as the ACL on the incoming profile blocks that IP.

The users in the direct should not register just to IP based auth, which the
CIDR attribute takes care of?

Just looks like it's not seeing the users in the directory at all, anything
I might be missing that just jumps right out?

Thanks
Ron

On Sun, Dec 28, 2008 at 4:20 PM, Brian West <brian at freeswitch.org> wrote:

>
> On Dec 28, 2008, at 12:52 PM, Ron McCarthy wrote:
>
> > Hi all,
> >
> > I would like to setup FS to have many context, basically we just
> > want to switch calls, and since we have been using Asterisk we want
> > to keep the context names the same, and well it's easier it seems to
> > me.
> >
> > This is for termination only, so just sending calls out, I would
> > think FS can do it the same way Asterisk does, where if the IP
> > matches then it will use those settings with that gateway and
> > diaplan since the context ties them all together. No matter what I
> > try the context always goes to default, I even took off all the
> > ACL's for internal and external profiles and tried gateways in each,
> > still no luck, i can't get around the default context. We are going
> > to be using IP based auth only, so no user/pass's ever.
>
>
> This type of authentication is still tied to the directory.  If you
> notice in the default config you'll notice the domains acl... that ACL
> is built off the cidr= attribute on the user tag in the directory.
>
>   <user id="1000" cidr="1.2.3.4/24">
>     <variables>
>       <variable name="user_context" value="my_context"/>
>     </variables>
>   </user>
>
> So now when you "reloadacl reloadxml" yes both together like that.
>
> I also just fixed a bug related to this due to recent directory layout
> changes.
>
> /b
>
> PS you can also set the context to "_domain_" and it'll auto on the
> from host.
>
>
> >
> >
> > Any help on this would be great, I have searched and searched and
> > everyone just uses the default context, but I am trying to avoid
> > that as the diaplan will get quite long as I will have to all kinds
> > of crazy matching, when a context is much simpler, and I would think
> > faster then using lots of dialplan logic.
> >
> > Thanks for any help on this!
> >
> > Ron
> > _____
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Dec 28 15:58:59 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Dec 2008 17:58:59 -0600
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
Message-ID: <171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>

I can bet you're on a rev before 10981 and after 10917... update you  
have a bug related to this.

/b

On Dec 28, 2008, at 5:54 PM, Ron McCarthy wrote:

> I see the "brian.xml" example has it, didn't check that one, whoops.
>
> Now I have added a user in the directory with the correct CIDR  
> attribute, yet when I send the call it seems to not use the  
> directory, the calls is coming from a Asterisk box, I have the  
> Asterisk box pointed to the "internal" profile's IP address, which I  
> assume the directory would use, and it gets rejected as the ACL on  
> the incoming profile blocks that IP.
>
> The users in the direct should not register just to IP based auth,  
> which the CIDR attribute takes care of?
>
> Just looks like it's not seeing the users in the directory at all,  
> anything I might be missing that just jumps right out?
>
> Thanks
> Ron



From ronmccar at gmail.com  Sun Dec 28 16:16:49 2008
From: ronmccar at gmail.com (Ron McCarthy)
Date: Sun, 28 Dec 2008 17:16:49 -0700
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
	<171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
Message-ID: <3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>

Wow, 10171 I guess Im way far behind! I am building right now, ill let ya
know if any issues, thanks again!



On Sun, Dec 28, 2008 at 4:58 PM, Brian West <brian at freeswitch.org> wrote:

> I can bet you're on a rev before 10981 and after 10917... update you
> have a bug related to this.
>
> /b
>
> On Dec 28, 2008, at 5:54 PM, Ron McCarthy wrote:
>
> > I see the "brian.xml" example has it, didn't check that one, whoops.
> >
> > Now I have added a user in the directory with the correct CIDR
> > attribute, yet when I send the call it seems to not use the
> > directory, the calls is coming from a Asterisk box, I have the
> > Asterisk box pointed to the "internal" profile's IP address, which I
> > assume the directory would use, and it gets rejected as the ACL on
> > the incoming profile blocks that IP.
> >
> > The users in the direct should not register just to IP based auth,
> > which the CIDR attribute takes care of?
> >
> > Just looks like it's not seeing the users in the directory at all,
> > anything I might be missing that just jumps right out?
> >
> > Thanks
> > Ron
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Dec 28 16:28:07 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Dec 2008 18:28:07 -0600
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
	<171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
	<3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>
Message-ID: <BC41ACAF-107F-4BDD-A6B9-D3824FFDD34B@freeswitch.org>

Wow then you didn't have this one bug.... but yah a little bit  
behind ... update just to be safe.  If you still have the problem  
please post.

/b

On Dec 28, 2008, at 6:16 PM, Ron McCarthy wrote:

> Wow, 10171 I guess Im way far behind! I am building right now, ill  
> let ya know if any issues, thanks again!



From ronmccar at gmail.com  Sun Dec 28 18:46:36 2008
From: ronmccar at gmail.com (Ron McCarthy)
Date: Sun, 28 Dec 2008 19:46:36 -0700
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <BC41ACAF-107F-4BDD-A6B9-D3824FFDD34B@freeswitch.org>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
	<171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
	<3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>
	<BC41ACAF-107F-4BDD-A6B9-D3824FFDD34B@freeswitch.org>
Message-ID: <3885f4fe0812281846g74d2febcs83cc33f8aaff05f4@mail.gmail.com>

Running 10981 now, same error.

I have:

  <user id="proxy-01" cidr="x.x.xx.x/32">
    <params>
      <param name="password" value="1"/>
    </params>
    <variables>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="newContext"/>
      <variable name="effective_caller_id_name" value="CID TEST"/>
      <variable name="effective_caller_id_number" value="1000"/>
    </variables>
  </user>

Anymore ideals?

Thansk


On Sun, Dec 28, 2008 at 5:28 PM, Brian West <brian at freeswitch.org> wrote:

> Wow then you didn't have this one bug.... but yah a little bit
> behind ... update just to be safe.  If you still have the problem
> please post.
>
> /b
>
> On Dec 28, 2008, at 6:16 PM, Ron McCarthy wrote:
>
> > Wow, 10171 I guess Im way far behind! I am building right now, ill
> > let ya know if any issues, thanks again!
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Dec 28 19:02:57 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Dec 2008 21:02:57 -0600
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <3885f4fe0812281846g74d2febcs83cc33f8aaff05f4@mail.gmail.com>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
	<171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
	<3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>
	<BC41ACAF-107F-4BDD-A6B9-D3824FFDD34B@freeswitch.org>
	<3885f4fe0812281846g74d2febcs83cc33f8aaff05f4@mail.gmail.com>
Message-ID: <99398E29-D7F6-4ED7-91AC-0064ACA8617A@freeswitch.org>

show me the output of reloadacl  , Chances are the domain in the  
acl.conf.xml and the one in your directory don't jive.

/b

On Dec 28, 2008, at 8:46 PM, Ron McCarthy wrote:

> Running 10981 now, same error.
>
> I have:
>
>   <user id="proxy-01" cidr="x.x.xx.x/32">
>     <params>
>       <param name="password" value="1"/>
>     </params>
>     <variables>
>       <variable name="accountcode" value="1000"/>
>       <variable name="user_context" value="newContext"/>
>       <variable name="effective_caller_id_name" value="CID TEST"/>
>       <variable name="effective_caller_id_number" value="1000"/>
>     </variables>
>   </user>
>
> Anymore ideals?
>
> Thansk



From yudha2008 at gmail.com  Mon Dec 29 04:38:47 2008
From: yudha2008 at gmail.com (Baskar)
Date: Mon, 29 Dec 2008 18:08:47 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
Message-ID: <d0640d380812290438x715f5294q199f481813c76f14@mail.gmail.com>

*Hi  **Michael,*


     Step i follow for the Tone Detect process







        Thanks for the Reply It is useful for me



On Sat, Dec 27, 2008 at 11:14 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Dec 27, 2008, at 5:56 AM, Baskar wrote:
>
> *Hi **Michael,*
> *
>     "I try to detect the tone before answering the call.
>
>      Is there any module for tone detect to be enabled"*
>
>
> I believe its in mod_dptools, if its not throwing an error that the
> application does not exist, then its fine.
>
>
>
> *     " I have set ignore_early_media=False **(False is case sensitive?)*"
>
>
> you can omit ignore_early_media, it defaults to false.
>
> *     But still no Tone is Detected.*
>
>
> Have you confirmed that tone frequency for your busy tone in your country
> and that the tone is actually being played as part of early media?
>
>
>
> Mike
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Warm Regards,
N.Baskar
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From yudha2008 at gmail.com  Mon Dec 29 05:04:43 2008
From: yudha2008 at gmail.com (Baskar)
Date: Mon, 29 Dec 2008 18:34:43 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
Message-ID: <d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>

*Hi  **Michael,*

*Steps I follow for the Tone Detect process*
*
**Step1: **From X-lite i called my no (eg: 1007==>9841799874 )
**Step2: Then i run the JavaScript in that also i have given same no
(9841799874)
* *Step3: While i run the JavaScript i should get the busy tone detect but i
cant ???*

*INTERNATIONAL TELECOMMUNICATION UNION given all national frequency *

*For India they have given*

*India (Republic of)
Acceptance tone -          400   1.0 on 4.0 off
Busy tone -                    400   0.75 on 0.75 off
Congestion tone -           400   0.25 on 0.25 off
Dial tone -                      400x25   continuous
Special dial tone -           400   2.8 on 0.2 off
Holding tone -                 400   0.25 on 0.25 off 0.25 on 3.25 off
Intrusion tone -               400   0.15 on 4.85 off
Refusal tone -                400   0.25 on 0.25 off
Ringing tone - I (local calls)          400x25   0.4 on 0.2 off 0.4 on 2..0
off
Ringing tone - II (NSD/ISD calls)   400x25   1.0 on 2.0 off
Route tone -                  400   0.1 on 0.9 off
Call waiting tone -          400   0.2 on 0.1 off 0.2 on 7.5 off *


*But in wiki.sangoma They have given frequency for india (openzap)*

*[in]
generate-dial => v=-7;%(1000,0,375,425)
detect-dial => 375,425
generate-ring => v=-7;%(2000,4000,440,480)
detect-ring => 440,480
generate-busy => v=-7;%(500,500,480,620)
detect-busy => 480,620
generate-attn => v=0;%(100,100,1400,2060,2450,2600)
detect-attn => 1400,2060,2450,2600
generate-callwaiting-sas => v=0;%(300,0,440)
detect-callwaiting-sas => 440
generate-callwaiting-cas => v=0;%(80,0,2750,2130)
detect-callwaiting-cas => 2750,2130
detect-fail1 => 913.8
detect-fail2 => 1370.6
detect-fail3 => 776.7
*

*
In INTERNATIONAL TELECOMMUNICATION UNION they have given busy tone frequency
is
Busy tone -                    400   0.75 on 0.75 off

But in wiki.sangoma They have given
generate-busy => v=-7;%(500,500,480,620)
detect-busy => 480,620*

*I have tried both 400 and 480,620 in JavaScript but still i cant detect
tone.*

*Guide me Which Frequency Should i use and Another thing What steps Should
be followed for the Tone Detect .
(i)Whether my step is correct .
           OR
(ii)Should i follow in different method.If so how?  Help To detect the tone
.....*

*
Thanks for the reply,

-- 
Warm Regards,
N.Baskar
*
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From fidibus83 at aol.com  Mon Dec 29 05:20:03 2008
From: fidibus83 at aol.com (fidibus83)
Date: Mon, 29 Dec 2008 14:20:03 +0100
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
Message-ID: <00a101c969b8$29c0f420$6445310a@Franzi>

Hello,

 

I want to getting mod xml cdr working, but when I start freeswitch I get
this Error:

 

[CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error
Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so

**/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**

 

Why can?t FS load the module?

 

Thanks!

 

Best regards
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From mike at jerris.com  Mon Dec 29 05:56:46 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Dec 2008 08:56:46 -0500
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
Message-ID: <AD38E1EF-851A-4407-9C02-63207C16BB8B@jerris.com>

I don't understand your steps.

On Dec 29, 2008, at 8:04 AM, Baskar <yudha2008 at gmail.com> wrote:

> Hi  Michael,
>
> Steps I follow for the Tone Detect process
>
> Step1: From X-lite i called my no (eg: 1007==>9841799874 )
> Step2: Then i run the JavaScript in that also i have given same no (9841799874 
> )
> Step3: While i run the JavaScript i should get the busy tone detect  
> but i cant ???
>
> INTERNATIONAL TELECOMMUNICATION UNION given all national frequency
>
> For India they have given
>
> India (Republic of)
> Acceptance tone -          400   1.0 on 4.0 off
> Busy tone -                    400   0.75 on 0.75 off
> Congestion tone -           400   0.25 on 0.25 off
> Dial tone -                      400x25   continuous
> Special dial tone -           400   2.8 on 0.2 off
> Holding tone -                 400   0.25 on 0.25 off 0.25 on 3.25 off
> Intrusion tone -               400   0.15 on 4.85 off
> Refusal tone -                400   0.25 on 0.25 off
> Ringing tone - I (local calls)          400x25   0.4 on 0.2 off 0.4  
> on 2..0 off
> Ringing tone - II (NSD/ISD calls)   400x25   1.0 on 2.0 off
> Route tone -                  400   0.1 on 0.9 off
> Call waiting tone -          400   0.2 on 0.1 off 0.2 on 7.5 off
>
>
> But in wiki.sangoma They have given frequency for india (openzap)
> [in]
> generate-dial => v=-7;%(1000,0,375,425)
>
> detect-dial => 375,425
> generate-ring => v=-7;%(2000,4000,440,480)
>
> detect-ring => 440,480
> generate-busy => v=-7;%(500,500,480,620)
>
> detect-busy => 480,620
> generate-attn => v=0;%(100,100,1400,2060,2450,2600)
>
> detect-attn => 1400,2060,2450,2600
> generate-callwaiting-sas => v=0;%(300,0,440)
>
> detect-callwaiting-sas => 440
> generate-callwaiting-cas => v=0;%(80,0,2750,2130)
>
> detect-callwaiting-cas => 2750,2130
> detect-fail1 => 913.8
>
> detect-fail2 => 1370.6
> detect-fail3 => 776.7
>
> In INTERNATIONAL TELECOMMUNICATION UNION they have given busy tone  
> frequency is
> Busy tone -                    400   0.75 on 0.75 off
>
> But in wiki.sangoma They have given
> generate-busy => v=-7;%(500,500,480,620)
> detect-busy => 480,620
>
> I have tried both 400 and 480,620 in JavaScript but still i cant  
> detect tone.
>
> Guide me Which Frequency Should i use and Another thing What steps  
> Should be followed for the Tone Detect .
> (i)Whether my step is correct .
>            OR
> (ii)Should i follow in different method.If so how?  Help To detect  
> the tone .....
>
>
> Thanks for the reply,
>
> -- 
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From wiltingtree at gmail.com  Mon Dec 29 06:28:30 2008
From: wiltingtree at gmail.com (Adam Wilt)
Date: Mon, 29 Dec 2008 09:28:30 -0500
Subject: [Freeswitch-users] onInputCallback unstable in Python
In-Reply-To: <eb7797d70812271815xdf10c72v5ea1b25098205f05@mail.gmail.com>
References: <eb7797d70812271815xdf10c72v5ea1b25098205f05@mail.gmail.com>
Message-ID: <eb7797d70812290628t1eae34f4wb19313c010a595b9@mail.gmail.com>

Should I add this to Jira?



On Sat, Dec 27, 2008 at 9:15 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> I'm using builds 10724 and 10914 to place an outbound call from the
> FreeSwitch console and use the onInputCallback functionality. My goal is to
> get mod_vmd working for me.
>
> When I run my script and press a touchtone to invoke the callback
> function,  I get some unstable behavior; sometimes it works fine, sometimes
> I get a core dump (send me an email at wiltingtree at gmail.com if anybody
> wants the core dump),  sometimes it gives me the following error:
>
>    TypeError: onInputCallback() takes exactly 3 arguments (0 given)
>
> In the documentation onInputCallback() takes 3 arguments, and I don't see
> how it would be very useful with zero.
>
> Here is a test script I put together which shows this behavior:
>
> import os
> from freeswitch import *
> def onInputCallback(session, what, obj):
>   consoleLog("INFO","IM IN THE CALLBACK!\n")
>   return("continue")
>
> def fsapi(session, stream, env, args):
>   consoleLog("INFO","Hello there!!!\n")
>   session =
> Session("{ignore_early_media=true}sofia/gateway/gafachi/1xxxxxxxxxx")
>   session.sleep(500)
>   session.setInputCallback(onInputCallback)
>   session.streamFile("/root/intro.wav")
>   consoleLog("info","Bye!\n")
>   session.hangup()
>   return(session)
>
>
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From msc at freeswitch.org  Mon Dec 29 07:40:06 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 29 Dec 2008 07:40:06 -0800
Subject: [Freeswitch-users] onInputCallback unstable in Python
In-Reply-To: <eb7797d70812290628t1eae34f4wb19313c010a595b9@mail.gmail.com>
References: <eb7797d70812271815xdf10c72v5ea1b25098205f05@mail.gmail.com>
	<eb7797d70812290628t1eae34f4wb19313c010a595b9@mail.gmail.com>
Message-ID: <C84D71BF-14A3-4DFF-AA82-1E9364004473@freeswitch.org>

A jira for the core dumps would be good, especially if you can  
reproduce the behavior.

Question: where does mod_vmd come into play?

-MC

Sent from my iPhone

On Dec 29, 2008, at 6:28 AM, "Adam Wilt" <wiltingtree at gmail.com> wrote:

> Should I add this to Jira?
>
>
>
> On Sat, Dec 27, 2008 at 9:15 PM, Adam Wilt <wiltingtree at gmail.com>  
> wrote:
> I'm using builds 10724 and 10914 to place an outbound call from the  
> FreeSwitch console and use the onInputCallback functionality. My  
> goal is to get mod_vmd working for me.
>
> When I run my script and press a touchtone to invoke the callback  
> function,  I get some unstable behavior; sometimes it works fine,  
> sometimes I get a core dump (send me an email at  
> wiltingtree at gmail.com if anybody wants the core dump),  sometimes it  
> gives me the following error:
>
>    TypeError: onInputCallback() takes exactly 3 arguments (0 given)
>
> In the documentation onInputCallback() takes 3 arguments, and I  
> don't see how it would be very useful with zero.
>
> Here is a test script I put together which shows this behavior:
>
> import os
> from freeswitch import *
> def onInputCallback(session, what, obj):
>   consoleLog("INFO","IM IN THE CALLBACK!\n")
>   return("continue")
>
> def fsapi(session, stream, env, args):
>   consoleLog("INFO","Hello there!!!\n")
>   session = Session("{ignore_early_media=true}sofia/gateway/gafachi/ 
> 1xxxxxxxxxx")
>   session.sleep(500)
>   session.setInputCallback(onInputCallback)
>   session.streamFile("/root/intro.wav")
>   consoleLog("info","Bye!\n")
>   session.hangup()
>   return(session)
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From intralanman at freeswitch.org  Mon Dec 29 07:42:50 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Mon, 29 Dec 2008 10:42:50 -0500
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <20081226000150.140990@gmx.net>
References: <20081225203811.79240@gmx.net>	<2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>
	<20081226000150.140990@gmx.net>
Message-ID: <4958EFFA.2080408@freeswitch.org>

can_man at gmx.de wrote:
> Hello,
>
> thank you for your answers. I am a step further now, it seems that just the "condition" tags as described in the wiki aren't enough. After sending the following xml I think I get stuck at the point Micheal mentioned:
>
>   
>> The $$ substitutions are only done in the static XML files.  Al
>>     
>
> FS complains that: Context default not found
>
>
> XML:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="RE Dial Plan For FreeSwitch">
> <context name="public">
Pay close attention here.... notice the context name in your XML and the 
context name that FreeSWITCH is saying it can't find....
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From msc at freeswitch.org  Mon Dec 29 07:45:27 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 29 Dec 2008 07:45:27 -0800
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
Message-ID: <2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>

For the sake of testing can you record a call that gets a busy signal?  
At least then we could analyze the audio and see what's going on.

If you need a dialplan example for this let me know.

-MC

Sent from my iPhone

On Dec 29, 2008, at 5:04 AM, Baskar <yudha2008 at gmail.com> wrote:

> Hi  Michael,
>
> Steps I follow for the Tone Detect process
>
> Step1: From X-lite i called my no (eg: 1007==>9841799874 )
> Step2: Then i run the JavaScript in that also i have given same no (9841799874 
> )
> Step3: While i run the JavaScript i should get the busy tone detect  
> but i cant ???
>
> INTERNATIONAL TELECOMMUNICATION UNION given all national frequency
>
> For India they have given
>
> India (Republic of)
> Acceptance tone -          400   1.0 on 4.0 off
> Busy tone -                    400   0.75 on 0.75 off
> Congestion tone -           400   0.25 on 0.25 off
> Dial tone -                      400x25   continuous
> Special dial tone -           400   2.8 on 0.2 off
> Holding tone -                 400   0.25 on 0.25 off 0.25 on 3.25 off
> Intrusion tone -               400   0.15 on 4.85 off
> Refusal tone -                400   0.25 on 0.25 off
> Ringing tone - I (local calls)          400x25   0.4 on 0.2 off 0.4  
> on 2..0 off
> Ringing tone - II (NSD/ISD calls)   400x25   1.0 on 2.0 off
> Route tone -                  400   0.1 on 0.9 off
> Call waiting tone -          400   0.2 on 0.1 off 0.2 on 7.5 off
>
>
> But in wiki.sangoma They have given frequency for india (openzap)
> [in]
> generate-dial => v=-7;%(1000,0,375,425)
>
> detect-dial => 375,425
> generate-ring => v=-7;%(2000,4000,440,480)
>
> detect-ring => 440,480
> generate-busy => v=-7;%(500,500,480,620)
>
> detect-busy => 480,620
> generate-attn => v=0;%(100,100,1400,2060,2450,2600)
>
> detect-attn => 1400,2060,2450,2600
> generate-callwaiting-sas => v=0;%(300,0,440)
>
> detect-callwaiting-sas => 440
> generate-callwaiting-cas => v=0;%(80,0,2750,2130)
>
> detect-callwaiting-cas => 2750,2130
> detect-fail1 => 913.8
>
> detect-fail2 => 1370.6
> detect-fail3 => 776.7
>
> In INTERNATIONAL TELECOMMUNICATION UNION they have given busy tone  
> frequency is
> Busy tone -                    400   0.75 on 0.75 off
>
> But in wiki.sangoma They have given
> generate-busy => v=-7;%(500,500,480,620)
> detect-busy => 480,620
>
> I have tried both 400 and 480,620 in JavaScript but still i cant  
> detect tone.
>
> Guide me Which Frequency Should i use and Another thing What steps  
> Should be followed for the Tone Detect .
> (i)Whether my step is correct .
>            OR
> (ii)Should i follow in different method.If so how?  Help To detect  
> the tone .....
>
>
> Thanks for the reply,
>
> -- 
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Mon Dec 29 07:50:42 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 29 Dec 2008 07:50:42 -0800
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <636b01c9691f$68802120$33014c0a@ws4>
References: <636b01c9691f$68802120$33014c0a@ws4>
Message-ID: <1F08393E-BC33-45C3-989D-6BAD3DC40963@freeswitch.org>

I wonder if putting a sleep statement in your shell script might help.  
If it's a timing issue then possibly the shell script is trying to  
access the file before FS and/or the OS are done with it. You would  
need to tinker with how long to sleep in order to find a value that  
works in all cases.

-MC

Sent from my iPhone

On Dec 28, 2008, at 11:06 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> Maybe I am going about this all wrong.  All I am trying to do is  
> process a recording file of a session after either one of the legs  
> hangs up and the call is over.  I am just trying to convert the wav  
> to mp3 and email it off.  So I have a bash script to do this.  The  
> dialplan is simple enought
>
>           <action application="set" data="RECORD_ANSWER_REQ=true"/>
>           <action application="set" data="RECORD_STEREO=true"/>
>           <action application="record_session" data="$${base_dir}/ 
> recordings/3329000_1.wav"/>
>           <action application="bridge" data="${enum_auto_route}"/>
>           <action application="hangup"/>
>
>
> using FS svn 10960
>
>
>
> But nothing I have tried seems to get it done.  I have tried to use  
> api_hangup_hook to call a lua script.  But the wav file is not yet  
> closed for some reason yet and I cannot seem to close it in the lua  
> script.  Basically I get the same problem if I use ?system? with  
> the api_hangup_hook to call the bash script to process the recording.
>
>
>
> I have tried to use the transfer application with the  
> api_hangup_hook to allow me to stop_record_session and then a system  
> call from another extension, but the transfer never happens on hangup.
>
>
>
> Am I missing a simple way to do this?  Is there something similar to  
> the ?h? extension in asterisk maybe?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From can_man at gmx.de  Mon Dec 29 08:31:52 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 29 Dec 2008 17:31:52 +0100
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <4958EFFA.2080408@freeswitch.org>
References: <20081225203811.79240@gmx.net>
	<2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>
	<20081226000150.140990@gmx.net> <4958EFFA.2080408@freeswitch.org>
Message-ID: <20081229163152.206580@gmx.net>

    
> >
> > FS complains that: Context default not found
> >
> >
> > XML:
> >
> > <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> > <document type="freeswitch/xml">
> > <section name="dialplan" description="RE Dial Plan For FreeSwitch">
> > <context name="public">

> Pay close attention here.... notice the context name in your XML and the 
> context name that FreeSWITCH is saying it can't find....

>From what I understand the forward rule for my external SIP number should be in context public and the internal "music on hold" should be in context default.

However, I got everything to work now by removing the crypto checks. It works when I reply:

    '''<?xml version="1.0" encoding="UTF-8" standalone="no"?>\n'''\
    '''<document type="freeswitch/xml">\n'''\
    '''<section name="dialplan" description="RE Dial Plan For FreeSwitch">\n'''\
    '''<context name="default">\n'''\
    '''<extension name="9999">\n'''\
    '''<condition field="destination_number" expression="^(9999)$">\n'''\
    '''<action application="answer"/>\n'''\
    '''<action application="playback" data="$${hold_music}"/>\n'''\
    '''</condition>\n'''\
    '''</extension>\n'''\
    '''</context>\n'''\
    '''</section>\n'''\
    '''</document>\n''' 

But I don't understand why it works with crypto checks on when I use the xml dial plan config files and not with xml curl. Anyway, I am happy that it works now and I can continue.

Thanks for your help.
Phil
-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From mike at jerris.com  Mon Dec 29 09:10:03 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Dec 2008 12:10:03 -0500
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
In-Reply-To: <00a101c969b8$29c0f420$6445310a@Franzi>
References: <00a101c969b8$29c0f420$6445310a@Franzi>
Message-ID: <018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com>

newer fedora (core 8 and later) have libcurl that is built against  
nspr for some reason, but we don't link against it.  if you configure  
freeswitch with --without-libcurl it will use our private copy instead  
of the distro copy and resolve this issue.

Mike

On Dec 29, 2008, at 8:20 AM, fidibus83 wrote:

> Hello,
>
> I want to getting mod xml cdr working, but when I start freeswitch I  
> get this Error:
>
> [CRIT] switch_loadable_module.c:756  
> switch_loadable_module_load_file() Error Loading module /usr/local/ 
> freeswitch/mod/mod_xml_cdr.so
> **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
>
> Why can?t FS load the module?
>
> Thanks!
>
> Best regards
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frank at impactfax.com  Mon Dec 29 09:39:47 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Mon, 29 Dec 2008 12:39:47 -0500
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <1F08393E-BC33-45C3-989D-6BAD3DC40963@freeswitch.org>
Message-ID: <6a2f01c969dc$71ccf970$33014c0a@ws4>

Yes.  I had tried that.  Put a sleep 15 in the shell script before I
looked at the file.  Same results however.  FS just does not appear to
be closing that record file on hangup.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins
Sent: Monday, December 29, 2008 10:51 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session_record post-processing
 
I wonder if putting a sleep statement in your shell script might help.
If it's a timing issue then possibly the shell script is trying to
access the file before FS and/or the OS are done with it. You would need
to tinker with how long to sleep in order to find a value that works in
all cases. 
 
-MC
 
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From msc at freeswitch.org  Mon Dec 29 11:55:33 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 11:55:33 -0800
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <6a2f01c969dc$71ccf970$33014c0a@ws4>
References: <1F08393E-BC33-45C3-989D-6BAD3DC40963@freeswitch.org>
	<6a2f01c969dc$71ccf970$33014c0a@ws4>
Message-ID: <87f2f3b90812291155h3104de95if9d78344aae933f8@mail.gmail.com>

Curious: what are your endpoints? Also, what codec(s), etc. are you using?
I'm using PCMU with openzap endpoints and I don't get anything like this at
all. I'd like to try and emulate what you've got more closely to see if I
can reproduce the symptoms.

Thanks,
MC

On Mon, Dec 29, 2008 at 9:39 AM, Frank @ Impact <frank at impactfax.com> wrote:

>  Yes.  I had tried that.  Put a sleep 15 in the shell script before I
> looked at the file.  Same results however.  FS just does not appear to be
> closing that record file on hangup.
>
>
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael S
> Collins
> *Sent:* Monday, December 29, 2008 10:51 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] session_record post-processing
>
>
>
> I wonder if putting a sleep statement in your shell script might help. If
> it's a timing issue then possibly the shell script is trying to access the
> file before FS and/or the OS are done with it. You would need to tinker with
> how long to sleep in order to find a value that works in all cases.
>
>
>
> -MC
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Mon Dec 29 14:46:22 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 29 Dec 2008 23:46:22 +0100
Subject: [Freeswitch-users] event_socket and stop_dtmf
Message-ID: <4959533E.5030708@gmx.net>

When I send a stop_dtmf command via event-socket, I get a
channel_execute and a channel_execute_complete message back. However FS
still accepts DTMFs and sends them via event-socket. In addition the
other party will hear the DTMF. So I expect the stop_dtmf command is not
really executed by FS.

Here is the message I send:
SendMsg
call-command: execute
execute-app-name: stop_dtmf
execute-app-arg: true
event-lock:true
I send this command while I deliver a number of announcements to the user

At Startup I get the following on the console
2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
switch_loadable_module_process() Adding Application 'stop_dtmf'
2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
switch_loadable_module_process() Adding Application 'stop_dtmf_generate'

When I push stop_dtmf I get the following
2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event()
sofia/internal/1005 at my.domain Command Execute stop_dtmf(true)

What am I doing wrong here?

Here's a console output when I push DTMF on either side after stop_dtmf
has been pushed and the 2 call legs are bridged.
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start
packet for [1] ts=417476340 dur=160/160/2000 seq=63290
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=320/320/2000 seq=63291
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=480/480/2000 seq=63292
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=640/640/2000 seq=63293
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=800/800/2000 seq=63294
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=960/960/2000 seq=63295
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=417476340 dur=2080/2080/2000 seq=63302
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=417476340 dur=2080/2080/2000 seq=63303
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=417476340 dur=2080/2080/2000 seq=63304
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start
packet for [1] ts=292343252 dur=160/160/2000 seq=53073
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=320/320/2000 seq=53074
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=480/480/2000 seq=53075
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=640/640/2000 seq=53076
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=800/800/2000 seq=53077
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=960/960/2000 seq=53078
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=292343252 dur=2080/2080/2000 seq=53085
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=292343252 dur=2080/2080/2000 seq=53086
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=292343252 dur=2080/2080/2000 seq=53087

I am also wondering why I receive multiple events (15) for each dtmf
pressed. I expect an echo floating back and forth and triggering dtmf, hein?

Best regards
Peter



From mike at jerris.com  Mon Dec 29 15:11:23 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Dec 2008 18:11:23 -0500
Subject: [Freeswitch-users] event_socket and stop_dtmf
In-Reply-To: <4959533E.5030708@gmx.net>
References: <4959533E.5030708@gmx.net>
Message-ID: <3A19067C-6627-493D-933E-948831F8F9C1@jerris.com>

stop_dtmf is JUST for the inband dtmf listener, I would guess you are  
getting dtmf via rfc2833 or some other method.  If you want to  
understand why we generate all those packets have a read of rfc 2833.


Mike

On Dec 29, 2008, at 5:46 PM, Peter P GMX wrote:

> When I send a stop_dtmf command via event-socket, I get a
> channel_execute and a channel_execute_complete message back. However  
> FS
> still accepts DTMFs and sends them via event-socket. In addition the
> other party will hear the DTMF. So I expect the stop_dtmf command is  
> not
> really executed by FS.
>
> Here is the message I send:
> SendMsg
> call-command: execute
> execute-app-name: stop_dtmf
> execute-app-arg: true
> event-lock:true
> I send this command while I deliver a number of announcements to the  
> user
>
> At Startup I get the following on the console
> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
> switch_loadable_module_process() Adding Application 'stop_dtmf'
> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
> switch_loadable_module_process() Adding Application  
> 'stop_dtmf_generate'
>
> When I push stop_dtmf I get the following
> 2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event()
> sofia/internal/1005 at my.domain Command Execute stop_dtmf(true)
>
> What am I doing wrong here?
>
> Here's a console output when I push DTMF on either side after  
> stop_dtmf
> has been pushed and the 2 call legs are bridged.
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start
> packet for [1] ts=417476340 dur=160/160/2000 seq=63290
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=320/320/2000 seq=63291
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=480/480/2000 seq=63292
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=640/640/2000 seq=63293
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=800/800/2000 seq=63294
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=960/960/2000 seq=63295
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=417476340 dur=2080/2080/2000 seq=63302
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=417476340 dur=2080/2080/2000 seq=63303
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=417476340 dur=2080/2080/2000 seq=63304
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start
> packet for [1] ts=292343252 dur=160/160/2000 seq=53073
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=320/320/2000 seq=53074
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=480/480/2000 seq=53075
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=640/640/2000 seq=53076
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=800/800/2000 seq=53077
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=960/960/2000 seq=53078
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=292343252 dur=2080/2080/2000 seq=53085
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=292343252 dur=2080/2080/2000 seq=53086
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=292343252 dur=2080/2080/2000 seq=53087
>
> I am also wondering why I receive multiple events (15) for each dtmf
> pressed. I expect an echo floating back and forth and triggering  
> dtmf, hein?
>
> Best regards
> Peter
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Prometheus001 at gmx.net  Mon Dec 29 16:41:54 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 30 Dec 2008 01:41:54 +0100
Subject: [Freeswitch-users] event_socket and stop_dtmf
In-Reply-To: <3A19067C-6627-493D-933E-948831F8F9C1@jerris.com>
References: <4959533E.5030708@gmx.net>
	<3A19067C-6627-493D-933E-948831F8F9C1@jerris.com>
Message-ID: <49596E52.1010400@gmx.net>

Yes, we get DTMF via rfc2833. If I set "dtmf-type" to "info" then
stop_dtmf works?

The reason why I want to suppress it, is the dtmf echoing (see end of my
mail). Do you see another way how I may suppress this dtmf echo? This is
severe sometimes and keeps ongoing for minutes under certain
circumstances. Thus further voice communication is no longer possible.

We have an incoming leg, play some announcements, build an outgoing leg
and then bridge those 2 channels. When any of the participiants then
pushes a dtmf, echoing begins.

Best regards
Peter

Michael Jerris schrieb:
> stop_dtmf is JUST for the inband dtmf listener, I would guess you are  
> getting dtmf via rfc2833 or some other method.  If you want to  
> understand why we generate all those packets have a read of rfc 2833.
>
>
> Mike
>
> On Dec 29, 2008, at 5:46 PM, Peter P GMX wrote:
>
>   
>> When I send a stop_dtmf command via event-socket, I get a
>> channel_execute and a channel_execute_complete message back. However  
>> FS
>> still accepts DTMFs and sends them via event-socket. In addition the
>> other party will hear the DTMF. So I expect the stop_dtmf command is  
>> not
>> really executed by FS.
>>
>> Here is the message I send:
>> SendMsg
>> call-command: execute
>> execute-app-name: stop_dtmf
>> execute-app-arg: true
>> event-lock:true
>> I send this command while I deliver a number of announcements to the  
>> user
>>
>> At Startup I get the following on the console
>> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
>> switch_loadable_module_process() Adding Application 'stop_dtmf'
>> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
>> switch_loadable_module_process() Adding Application  
>> 'stop_dtmf_generate'
>>
>> When I push stop_dtmf I get the following
>> 2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event()
>> sofia/internal/1005 at my.domain Command Execute stop_dtmf(true)
>>
>> What am I doing wrong here?
>>
>> Here's a console output when I push DTMF on either side after  
>> stop_dtmf
>> has been pushed and the 2 call legs are bridged.
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start
>> packet for [1] ts=417476340 dur=160/160/2000 seq=63290
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=320/320/2000 seq=63291
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=480/480/2000 seq=63292
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=640/640/2000 seq=63293
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=800/800/2000 seq=63294
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=960/960/2000 seq=63295
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=417476340 dur=2080/2080/2000 seq=63302
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=417476340 dur=2080/2080/2000 seq=63303
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=417476340 dur=2080/2080/2000 seq=63304
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start
>> packet for [1] ts=292343252 dur=160/160/2000 seq=53073
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=320/320/2000 seq=53074
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=480/480/2000 seq=53075
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=640/640/2000 seq=53076
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=800/800/2000 seq=53077
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=960/960/2000 seq=53078
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=292343252 dur=2080/2080/2000 seq=53085
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=292343252 dur=2080/2080/2000 seq=53086
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=292343252 dur=2080/2080/2000 seq=53087
>>
>> I am also wondering why I receive multiple events (15) for each dtmf
>> pressed. I expect an echo floating back and forth and triggering  
>> dtmf, hein?
>>
>> Best regards
>> Peter
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From mike at jerris.com  Mon Dec 29 17:00:20 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Dec 2008 20:00:20 -0500
Subject: [Freeswitch-users] event_socket and stop_dtmf
In-Reply-To: <49596E52.1010400@gmx.net>
References: <4959533E.5030708@gmx.net>
	<3A19067C-6627-493D-933E-948831F8F9C1@jerris.com>
	<49596E52.1010400@gmx.net>
Message-ID: <0CC84B14-4BB8-48B2-8D87-A6C953624AF5@jerris.com>



On Dec 29, 2008, at 7:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Yes, we get DTMF via rfc2833. If I set "dtmf-type" to "info" then
> stop_dtmf works?

No, it is ONLY for inband

>
> The reason why I want to suppress it, is the dtmf echoing (see end  
> of my
> mail)

This just shows sending dtmf, not any echo.


> Do you see another way how I may suppress this dtmf echo? This is
> severe sometimes and keeps ongoing for minutes under certain
> circumstances. Thus further voice communication is no longer possible.
>

Your not getting this from dtmf in 2833.  If you have this issue you  
will probably need to fix it at the point where the dtmf is converted  
to imband.

> We have an incoming leg, play some announcements, build an outgoing  
> leg
> and then bridge those 2 channels. When any of the participiants then
> pushes a dtmf, echoing begins.
>
> Best regards
> Peter

This sounds like a Very broken provider and you should be asking them  
to correct this echo issue.

Mike
>
>
> Michael Jerris schrieb:
>> stop_dtmf is JUST for the inband dtmf listener, I would guess you are
>> getting dtmf via rfc2833 or some other method.  If you want to
>> understand why we generate all those packets have a read of rfc 2833.
>>
>>
>> Mike
>>
>> On Dec 29, 2008, at 5:46 PM, Peter P GMX wrote:
>>
>>
>>> When I send a stop_dtmf command via event-socket, I get a
>>> channel_execute and a channel_execute_complete message back. However
>>> FS
>>> still accepts DTMFs and sends them via event-socket. In addition the
>>> other party will hear the DTMF. So I expect the stop_dtmf command is
>>> not
>>> really executed by FS.
>>>
>>> Here is the message I send:
>>> SendMsg
>>> call-command: execute
>>> execute-app-name: stop_dtmf
>>> execute-app-arg: true
>>> event-lock:true
>>> I send this command while I deliver a number of announcements to the
>>> user
>>>
>>> At Startup I get the following on the console
>>> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
>>> switch_loadable_module_process() Adding Application 'stop_dtmf'
>>> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
>>> switch_loadable_module_process() Adding Application
>>> 'stop_dtmf_generate'
>>>
>>> When I push stop_dtmf I get the following
>>> 2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391  
>>> switch_ivr_parse_event()
>>> sofia/internal/1005 at my.domain Command Execute stop_dtmf(true)
>>>
>>> What am I doing wrong here?
>>>
>>> Here's a console output when I push DTMF on either side after
>>> stop_dtmf
>>> has been pushed and the 2 call legs are bridged.
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start
>>> packet for [1] ts=417476340 dur=160/160/2000 seq=63290
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=320/320/2000 seq=63291
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=480/480/2000 seq=63292
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=640/640/2000 seq=63293
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=800/800/2000 seq=63294
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=960/960/2000 seq=63295
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=417476340 dur=2080/2080/2000 seq=63302
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=417476340 dur=2080/2080/2000 seq=63303
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=417476340 dur=2080/2080/2000 seq=63304
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start
>>> packet for [1] ts=292343252 dur=160/160/2000 seq=53073
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=320/320/2000 seq=53074
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=480/480/2000 seq=53075
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=640/640/2000 seq=53076
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=800/800/2000 seq=53077
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=960/960/2000 seq=53078
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=292343252 dur=2080/2080/2000 seq=53085
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=292343252 dur=2080/2080/2000 seq=53086
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=292343252 dur=2080/2080/2000 seq=53087
>>>
>>> I am also wondering why I receive multiple events (15) for each dtmf
>>> pressed. I expect an echo floating back and forth and triggering
>>> dtmf, hein?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Mon Dec 29 17:19:44 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 17:19:44 -0800
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client Available!
Message-ID: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>

Hello FreeSWITCHers!

Just a heads up, there are lots of cool things happening with FreeSWITCH.
Please check out the latest here:

http://freeswitch.org/node/156

Stay tuned for more news from the FreeSWITCH camp.

-MC (mercutioviz)
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From klaus.teller at gmx.net  Mon Dec 29 17:39:38 2008
From: klaus.teller at gmx.net (Klaus Teller)
Date: Tue, 30 Dec 2008 02:39:38 +0100
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
 Available!
In-Reply-To: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
Message-ID: <20081230013938.184760@gmx.net>

This is a very much awaited tool. Thanks to you guys. Sounds like 2009 will be a very exciting year in the community.

Klaus.
-------- Original-Nachricht --------
> Datum: Mon, 29 Dec 2008 17:19:44 -0800
> Von: "Michael Collins" <msc at freeswitch.org>
> An: freeswitch-users at lists.freeswitch.org, freeswitch-dev at lists.freeswitch.org
> Betreff: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client Available!

> Hello FreeSWITCHers!
> 
> Just a heads up, there are lots of cool things happening with FreeSWITCH.
> Please check out the latest here:
> 
> http://freeswitch.org/node/156
> 
> Stay tuned for more news from the FreeSWITCH camp.
> 
> -MC (mercutioviz)

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From jason at jasonjgw.net  Mon Dec 29 18:19:42 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 30 Dec 2008 13:19:42 +1100
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH
	Client	Available!
In-Reply-To: <20081230013938.184760@gmx.net>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
	<20081230013938.184760@gmx.net>
Message-ID: <20081230021942.GA31689@jdc.jasonjgw.net>

Klaus Teller <klaus.teller at gmx.net> wrote:
> This is a very much awaited tool. Thanks to you guys. Sounds like 2009 will
> be a very exciting year in the community.

I agree.

Thanks are due to the developers for this excellent work.

I compiled it, copied fs_cli to /usr/local/bin, and now:
jason at jdc:~$ fs_cli
freeswitch at default>

Thanks!



From jason at jasonjgw.net  Mon Dec 29 20:00:49 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 30 Dec 2008 15:00:49 +1100
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH
	Client	Available!
In-Reply-To: <20081230021942.GA31689@jdc.jasonjgw.net>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
	<20081230013938.184760@gmx.net>
	<20081230021942.GA31689@jdc.jasonjgw.net>
Message-ID: <20081230040049.GA2409@jdc.jasonjgw.net>

By the way, the command to exit fs_cli is /exit (or /bye or /quit).

Commands starting with / are handled internally by the process_command()
function of the CLI, instead of being treated as FreeSWITCH API commands.



From yudha2008 at gmail.com  Mon Dec 29 22:45:06 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 30 Dec 2008 12:15:06 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
	<2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>
Message-ID: <d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>

Hi Michael Jerris,

I will explain what i am currently doing :I don't understand

Step 1: From the xlite phone I have dialed a number and we were on the
conversation with one extension (1007 is my extension and my mobile No
9841799874)

Step 2: From the freeswitch console I am executing a javascript file with
tone detect like the one below,

My JavaScript :

session1 = new Session();

session1.originate(session1,"{ignore_early_media=false}sofia/internal/
1003 at 172.20.201.67");

session1.execute("tone_detect","busy 480,620 r +30 transfer '1000' 3");

session1.execute("bridge", "sofia/default/9841799874 at 172.20.191.228");

session1.execute("transfer", "9841799874");

session1.hangup;
            In the above script and in step1 the telephone numbers are same;
since the script is not detecting that the phone number is busy.



Hi Michael S Collins,

            Please let me know the script, so that it would be helpful for
me.

Thanks for the Reply,

Warm Regards,
N.Baskar
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From msc at freeswitch.org  Mon Dec 29 22:53:42 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 22:53:42 -0800
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <20081230040049.GA2409@jdc.jasonjgw.net>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
	<20081230013938.184760@gmx.net>
	<20081230021942.GA31689@jdc.jasonjgw.net>
	<20081230040049.GA2409@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812292253t23a67b1o784439714f0db595@mail.gmail.com>

Don't forget that there's a nice wiki page for fs_cli:
http://wiki.freeswitch.org/wiki/Fs_cli

-MC

On Dec 29, 2008, at 8:00 PM, Jason White <jason at jasonjgw.net> wrote:

> By the way, the command to exit fs_cli is /exit (or /bye or /quit).
>
> Commands starting with / are handled internally by the
> process_command()
> function of the CLI, instead of being treated as FreeSWITCH API
> commands.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From krice at suspicious.org  Mon Dec 29 22:57:45 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 30 Dec 2008 00:57:45 -0600
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
 Available!
In-Reply-To: <20081230040049.GA2409@jdc.jasonjgw.net>
Message-ID: <C57F2289.4B7B7%krice@suspicious.org>

You can also use the ... Command to exit fs_cli and there are a few more
commands that are locally processed. (note: on the FS main console ... Will
cause fs to shutdown. Fs_cli interprets this locally and it does not shut
down the main system. You stll need to do fsclt shutdown or something
similar)

See the wiki for more information M Collins did a pretty good job
documenting it

K


> From: Jason White <jason at jasonjgw.net>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 30 Dec 2008 15:00:49 +1100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
> Available!
> 
> By the way, the command to exit fs_cli is /exit (or /bye or /quit).
> 
> Commands starting with / are handled internally by the process_command()
> function of the CLI, instead of being treated as FreeSWITCH API commands.
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From msc at freeswitch.org  Mon Dec 29 23:21:51 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 23:21:51 -0800
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
	<2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>
	<d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>
Message-ID: <87f2f3b90812292321nfae4d0ckdc98106bf583c07@mail.gmail.com>

On Mon, Dec 29, 2008 at 10:45 PM, Baskar <yudha2008 at gmail.com> wrote:

> Hi Michael Jerris,
>
>  I will explain what i am currently doing : I don't understand
>
> Step 1: From the xlite phone I have dialed a number and we were on the
> conversation with one extension (1007 is my extension and my mobile No
> 9841799874)
>
> Step 2: From the freeswitch console I am executing a javascript file with
> tone detect like the one below,
>
> My JavaScript :
>
> session1 = new Session();
>
> session1.originate(session1,"{ignore_early_media=false}sofia/internal/
> 1003 at 172.20.201.67");
>
> session1.execute("tone_detect","busy 480,620 r +30 transfer '1000' 3");
>
I think the above line is part of the problem. The "+30" literally means
'watch for these tones for 30 milliseconds, and then don't watch any more.'
I think what you want here is +30000. Can you try that and see if there's a
difference?


> session1.execute("bridge", "sofia/default/9841799874 at 172.20.191.228");
>
> session1.execute("transfer", "9841799874");
>
> session1.hangup;
>             In the above script and in step1 the telephone numbers are
> same; since the script is not detecting that the phone number is busy.
>
>
>
> Hi Michael S Collins,
>
>             Please let me know the script, so that it would be helpful for
> me.
>
Here's a simple extension that I use for recording. It uses the uuid of the
call for the file name. I don't know if the "pre_answer" application is
absolutely necessary or not, but I do it just to be certain that I get the
early media.

<extension name="record early media">
  <condition field="destination_number" expression="^82(\d+)$"> <!-- 82 +
phone number -->
      <action application="bridge" data="sofia/default/$1 at 172.20.191.228"/>
      <!-- we can skip all this early media stuff if the far end answers -->
      <action application="set" data="execute_on_answer=transfer ANSWERCALL
XML default"/>
      <action application="sleep" data="100"/>
      <action application="pre_answer"/>
      <action application="record_session" data="/tmp/${uuid}__1.wav"/>
      <action application="sleep" data="500"/>
      <!-- If call gets here then probably we're in progress media; get
ready to handle various tones -->
      <!-- Look for busy and SIT tones right away, then wait a bit before
looking for a ring signal -->
      <action application="tone_detect" data="BUSY 480,620 r +25000 set
BUSY=true"/>  <!-- detect busy -->
      <action application="tone_detect" data="SIT1LO 913.8 r +25000 set
SIT1LO=true"/>   <!--1st segment low -->
      <!-- ... look for other SIT tones if you need to; snipped from this
example -->
      <!-- Now sleep while other end is ringing -->
      <action application="sleep" data="25010"/>
      <!-- if far end answers then we never get here...-->
      <!-- Use Lua script to check value of BUSY, SIT1LO, etc. -->
      <!-- If !BUSY and !SIT1LO, etc. then call is a no answer -->
      <!-- I use extensions for each scenario, e.g.
        <condition field="destination_number" expression="ANSWERCALL">
        OR
        <condition field="destination_number" expression="BUSYCALL">
        etc.
      -->
  </condition>
</extension>

In any case, if you call a busy number it should record the early media. You
can change the sleep time from 25010 to a shorter duration, like maybe
10000. You could also insert an info app to dump the channel variables and
see if tone_detect set your channel variable(s).

Hope this helps!
-MC

> Thanks for the Reply,
>
> Warm Regards,
> N.Baskar
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Dec 29 23:24:22 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 23:24:22 -0800
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <C57F2289.4B7B7%krice@suspicious.org>
References: <20081230040049.GA2409@jdc.jasonjgw.net>
	<C57F2289.4B7B7%krice@suspicious.org>
Message-ID: <87f2f3b90812292324r1c070458sb20c96db5df2be53@mail.gmail.com>

Ken,

Thanks for the clarification. I will make a note of this in the wiki. Also,
can you hum a few bars and tell us what the "/filter" command does?

-MC

On Mon, Dec 29, 2008 at 10:57 PM, Ken Rice <krice at suspicious.org> wrote:

> You can also use the ... Command to exit fs_cli and there are a few more
> commands that are locally processed. (note: on the FS main console ... Will
> cause fs to shutdown. Fs_cli interprets this locally and it does not shut
> down the main system. You stll need to do fsclt shutdown or something
> similar)
>
> See the wiki for more information M Collins did a pretty good job
> documenting it
>
> K
>
>
> > From: Jason White <jason at jasonjgw.net>
> > Reply-To: <freeswitch-users at lists.freeswitch.org>
> > Date: Tue, 30 Dec 2008 15:00:49 +1100
> > To: <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
> > Available!
> >
> > By the way, the command to exit fs_cli is /exit (or /bye or /quit).
> >
> > Commands starting with / are handled internally by the process_command()
> > function of the CLI, instead of being treated as FreeSWITCH API commands.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fidibus83 at aol.com  Tue Dec 30 01:46:17 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 30 Dec 2008 10:46:17 +0100
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
In-Reply-To: <018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com>
References: <00a101c969b8$29c0f420$6445310a@Franzi>
	<018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com>
Message-ID: <008a01c96a63$76c35870$6445310a@Franzi>

Thanks for your answer. But I don?t know what I have to do now. I?m a newbie
in FS.  How do I configure FS without libcurl?

 

Thanks, fidibus

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael Jerris
Gesendet: Montag, 29. Dezember 2008 18:10
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Can't load module mod_xml_cdr.so

 

newer fedora (core 8 and later) have libcurl that is built against nspr for
some reason, but we don't link against it.  if you configure freeswitch with
--without-libcurl it will use our private copy instead of the distro copy
and resolve this issue.

 

Mike

 

On Dec 29, 2008, at 8:20 AM, fidibus83 wrote:





Hello,

 

I want to getting mod xml cdr working, but when I start freeswitch I get
this Error:

 

[CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error
Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so

**/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**

 

Why can?t FS load the module?

 

Thanks!

 

Best regards

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

=
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From jason at jasonjgw.net  Tue Dec 30 01:59:50 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 30 Dec 2008 20:59:50 +1100
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
In-Reply-To: <008a01c96a63$76c35870$6445310a@Franzi>
References: <00a101c969b8$29c0f420$6445310a@Franzi>
	<018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com>
	<008a01c96a63$76c35870$6445310a@Franzi>
Message-ID: <20081230095950.GA5884@jdc.jasonjgw.net>

fidibus83 <fidibus83 at aol.com> wrote:
> Thanks for your answer. But I don?t know what I have to do now. I?m a
> newbie in FS.  How do I configure FS without libcurl?

./configure --without-libcurl
make
(then as root)
make install

In other words, run the configure script with the --without-libcurl option,
then recompile FreeSWITCH.



From fidibus83 at aol.com  Tue Dec 30 02:38:38 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 30 Dec 2008 11:38:38 +0100
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
In-Reply-To: <20081230095950.GA5884@jdc.jasonjgw.net>
References: <00a101c969b8$29c0f420$6445310a@Franzi><018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com><008a01c96a63$76c35870$6445310a@Franzi>
	<20081230095950.GA5884@jdc.jasonjgw.net>
Message-ID: <00b001c96a6a$c72fdc50$6445310a@Franzi>

Thanks. The Error is removed.

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jason
White
Gesendet: Dienstag, 30. Dezember 2008 11:00
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Can't load module mod_xml_cdr.so

fidibus83 <fidibus83 at aol.com> wrote:
> Thanks for your answer. But I don?t know what I have to do now. I?m a
> newbie in FS.  How do I configure FS without libcurl?

./configure --without-libcurl
make
(then as root)
make install

In other words, run the configure script with the --without-libcurl option,
then recompile FreeSWITCH.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From ivan at myrvold.org  Tue Dec 30 03:03:04 2008
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 30 Dec 2008 12:03:04 +0100
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <87f2f3b90812292324r1c070458sb20c96db5df2be53@mail.gmail.com>
References: <20081230040049.GA2409@jdc.jasonjgw.net>
	<C57F2289.4B7B7%krice@suspicious.org>
	<87f2f3b90812292324r1c070458sb20c96db5df2be53@mail.gmail.com>
Message-ID: <B95277C3-C7E9-43A7-B4F4-6B55B7BD17CE@myrvold.org>

I found out that both "/event" and "/events" worked as commands, but  
only "/noevents" worked, not "/noevent", although the Wiki says "/ 
noevent".

Ivan

Den 30. des.. 2008 kl. 08:24 skrev Michael Collins:

> Ken,
>
> Thanks for the clarification. I will make a note of this in the  
> wiki. Also, can you hum a few bars and tell us what the "/filter"  
> command does?
>
> -MC
>
> On Mon, Dec 29, 2008 at 10:57 PM, Ken Rice <krice at suspicious.org>  
> wrote:
> You can also use the ... Command to exit fs_cli and there are a few  
> more
> commands that are locally processed. (note: on the FS main  
> console ... Will
> cause fs to shutdown. Fs_cli interprets this locally and it does not  
> shut
> down the main system. You stll need to do fsclt shutdown or something
> similar)
>
> See the wiki for more information M Collins did a pretty good job
> documenting it
>
> K
>
>
> > From: Jason White <jason at jasonjgw.net>
> > Reply-To: <freeswitch-users at lists.freeswitch.org>
> > Date: Tue, 30 Dec 2008 15:00:49 +1100
> > To: <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH  
> Client
> > Available!
> >
> > By the way, the command to exit fs_cli is /exit (or /bye or /quit).
> >
> > Commands starting with / are handled internally by the  
> process_command()
> > function of the CLI, instead of being treated as FreeSWITCH API  
> commands.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kawarod at laposte.net  Tue Dec 30 04:21:32 2008
From: kawarod at laposte.net (rod)
Date: Tue, 30 Dec 2008 16:21:32 +0400
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
Message-ID: <495A124C.3040006@laposte.net>

Hi all,

I know that freeswitch has not been designed as a pure sip 
proxy/registrar, but I'm wondering how many subscribers could be handled 
by FS.

I setup the following test environment:
    - Kamailio 1.4.2 as the registrar
    - all invite requests are flowing through FS, even for a call 
between 2 registered subscribers. Many reasons for this: the calls CDR 
are centralized in the same format, I can easily add a billing ID to a 
call, proceed to recording, set the caller as anonymous if requested...
    - FS is used also as a SBC
   
There is still a lot of work to do, mainly on the call forwarding 
feature and this is why I'm wondering (simply out of curiosity) what 
could have been achieved using only FS (easier to setup when only one 
equipment is involved :) ).

I'd like to register 40 000 subscribers (if each user registers every 
60s, you have approx 670 registration per second, this setup is working 
on Kamailio).

I did the following to increase FS performance regarding registration:
    - put the directory containing users in a RAMDISK
    - put the db directory in a RAMDISK

with this I was able to reach 190 registration per second (50 without 
the ramdisk) but for one SIP account, not too useful :p (for your 
information I see a huge improvement when switching from 1.0.1 phoenix: 
150cps to FS svn 105xx: 190)
When trying with 25000 SIP accounts, I got no more than 30cps.

Then I tried to use the odbc mysql for registration, using this I was 
able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these 
tests, the presence support has been disabled.

As the IO performance seems to be a bottleneck, I'd like to know if 
there is a way to store the registration in memory only without database 
persistency.

This thread is there only to share tips, not to complain about FS poor 
performance as a SIP registrar when compared to Kamailio. If I compare 
FS to a commercial SBC I'm using in production, I have to say that FS is 
really a great piece of software (lacks only statistics module, snmp, 
and heartbeat redundancy for failover).

regards,
rod











From dyfet at gnutelephony.org  Tue Dec 30 05:07:57 2008
From: dyfet at gnutelephony.org (David Sugar)
Date: Tue, 30 Dec 2008 08:07:57 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A124C.3040006@laposte.net>
References: <495A124C.3040006@laposte.net>
Message-ID: <495A1D2D.3070507@gnutelephony.org>

You actually have potentially ~1320 effective "SIP transactions" per
second to support 40000 registered ua's with a 60s refresh.  This is
because the ua sends it's registration refresh unauthenticated.  The
registrar will then push back an authentication challenge request so the
ua can prove its identity, at which point the ua then repeats the same
transaction, but with authentication credentials attached.

rod wrote:
> Hi all,
> 
> I know that freeswitch has not been designed as a pure sip 
> proxy/registrar, but I'm wondering how many subscribers could be handled 
> by FS.
> 
> I setup the following test environment:
>     - Kamailio 1.4.2 as the registrar
>     - all invite requests are flowing through FS, even for a call 
> between 2 registered subscribers. Many reasons for this: the calls CDR 
> are centralized in the same format, I can easily add a billing ID to a 
> call, proceed to recording, set the caller as anonymous if requested...
>     - FS is used also as a SBC
>    
> There is still a lot of work to do, mainly on the call forwarding 
> feature and this is why I'm wondering (simply out of curiosity) what 
> could have been achieved using only FS (easier to setup when only one 
> equipment is involved :) ).
> 
> I'd like to register 40 000 subscribers (if each user registers every 
> 60s, you have approx 670 registration per second, this setup is working 
> on Kamailio).
> 
> I did the following to increase FS performance regarding registration:
>     - put the directory containing users in a RAMDISK
>     - put the db directory in a RAMDISK
> 
> with this I was able to reach 190 registration per second (50 without 
> the ramdisk) but for one SIP account, not too useful :p (for your 
> information I see a huge improvement when switching from 1.0.1 phoenix: 
> 150cps to FS svn 105xx: 190)
> When trying with 25000 SIP accounts, I got no more than 30cps.
> 
> Then I tried to use the odbc mysql for registration, using this I was 
> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these 
> tests, the presence support has been disabled.
> 
> As the IO performance seems to be a bottleneck, I'd like to know if 
> there is a way to store the registration in memory only without database 
> persistency.
> 
> This thread is there only to share tips, not to complain about FS poor 
> performance as a SIP registrar when compared to Kamailio. If I compare 
> FS to a commercial SBC I'm using in production, I have to say that FS is 
> really a great piece of software (lacks only statistics module, snmp, 
> and heartbeat redundancy for failover).
> 
> regards,
> rod
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gmaruzz at celliax.org  Tue Dec 30 05:27:36 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 30 Dec 2008 14:27:36 +0100
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A1D2D.3070507@gnutelephony.org>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
Message-ID: <7b197bef0812300527x5073b212j38b1a60f475440f6@mail.gmail.com>

Hi David,

very happy to read you on the FS list!

We met in 2001 at OSCon San Diego, where you "infected" me with the
telephony virus :-).

You did great work with the Bayonne project, really breaking new ground.

Thank you,

happy hacking,

happy new year!!!!



Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Tue, Dec 30, 2008 at 2:07 PM, David Sugar <dyfet at gnutelephony.org> wrote:
> You actually have potentially ~1320 effective "SIP transactions" per
> second to support 40000 registered ua's with a 60s refresh.  This is
> because the ua sends it's registration refresh unauthenticated.  The
> registrar will then push back an authentication challenge request so the
> ua can prove its identity, at which point the ua then repeats the same
> transaction, but with authentication credentials attached.
>
> rod wrote:
>> Hi all,
>>
>> I know that freeswitch has not been designed as a pure sip
>> proxy/registrar, but I'm wondering how many subscribers could be handled
>> by FS.
>>
>> I setup the following test environment:
>>     - Kamailio 1.4.2 as the registrar
>>     - all invite requests are flowing through FS, even for a call
>> between 2 registered subscribers. Many reasons for this: the calls CDR
>> are centralized in the same format, I can easily add a billing ID to a
>> call, proceed to recording, set the caller as anonymous if requested...
>>     - FS is used also as a SBC
>>
>> There is still a lot of work to do, mainly on the call forwarding
>> feature and this is why I'm wondering (simply out of curiosity) what
>> could have been achieved using only FS (easier to setup when only one
>> equipment is involved :) ).
>>
>> I'd like to register 40 000 subscribers (if each user registers every
>> 60s, you have approx 670 registration per second, this setup is working
>> on Kamailio).
>>
>> I did the following to increase FS performance regarding registration:
>>     - put the directory containing users in a RAMDISK
>>     - put the db directory in a RAMDISK
>>
>> with this I was able to reach 190 registration per second (50 without
>> the ramdisk) but for one SIP account, not too useful :p (for your
>> information I see a huge improvement when switching from 1.0.1 phoenix:
>> 150cps to FS svn 105xx: 190)
>> When trying with 25000 SIP accounts, I got no more than 30cps.
>>
>> Then I tried to use the odbc mysql for registration, using this I was
>> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these
>> tests, the presence support has been disabled.
>>
>> As the IO performance seems to be a bottleneck, I'd like to know if
>> there is a way to store the registration in memory only without database
>> persistency.
>>
>> This thread is there only to share tips, not to complain about FS poor
>> performance as a SIP registrar when compared to Kamailio. If I compare
>> FS to a commercial SBC I'm using in production, I have to say that FS is
>> really a great piece of software (lacks only statistics module, snmp,
>> and heartbeat redundancy for failover).
>>
>> regards,
>> rod
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From peder at networkoblivion.com  Tue Dec 30 05:32:16 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Tue, 30 Dec 2008 07:32:16 -0600
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A1D2D.3070507@gnutelephony.org>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
Message-ID: <495A22E0.3040904@networkoblivion.com>

>  This is
> because the ua sends it's registration refresh unauthenticated.  The
> registrar will then push back an authentication challenge request so the
> ua can prove its identity, at which point the ua then repeats the same
> transaction, but with authentication credentials attached.

Why does it do that?  Every time I do a debug, I see the first request 
denied as unauthorized and then it always comes right back and gets 
registered ok.  Is it part of the SIP spec to try unauthenticated first? 
    I would think you could set something on the UA to cut out the extra 
traffic.



From peder at networkoblivion.com  Tue Dec 30 05:37:07 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Tue, 30 Dec 2008 07:37:07 -0600
Subject: [Freeswitch-users] Register Interval
Message-ID: <495A2403.3090706@networkoblivion.com>

What do most people use as a register interval for phones?  On *, we 
always used 5 minutes and then had qualify setup, so we could keep track 
of the phones on a per minute basis as it "pinged" them every minute. 
FS doesn't do this to phones unless they are NAT'd, so if the reg is 5, 
we don't get any update for 5 minutes.  I like the idea of checking my 
phones every minute so that I know if there is a problem right away when 
someone calls with an issue, rather than having to wait up to 5 minutes 
to see if it is still alive.

Are most people using a small interval like 60 seconds?  Or do they set 
it longer and just assume the phones are still alive in between 
registrations?


Peder


From gmaruzz at celliax.org  Tue Dec 30 05:38:28 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 30 Dec 2008 14:38:28 +0100
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A22E0.3040904@networkoblivion.com>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
	<495A22E0.3040904@networkoblivion.com>
Message-ID: <7b197bef0812300538p48324e5ape8cfb1d50599e3fa@mail.gmail.com>

Yes, it is part of the SIP specs.
BTW, also HTTP works the same way.

Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Tue, Dec 30, 2008 at 2:32 PM, peder at networkoblivion.com
<peder at networkoblivion.com> wrote:
>>  This is
>> because the ua sends it's registration refresh unauthenticated.  The
>> registrar will then push back an authentication challenge request so the
>> ua can prove its identity, at which point the ua then repeats the same
>> transaction, but with authentication credentials attached.
>
> Why does it do that?  Every time I do a debug, I see the first request
> denied as unauthorized and then it always comes right back and gets
> registered ok.  Is it part of the SIP spec to try unauthenticated first?
>    I would think you could set something on the UA to cut out the extra
> traffic.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mgg at giagnocavo.net  Tue Dec 30 05:54:12 2008
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 30 Dec 2008 08:54:12 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A22E0.3040904@networkoblivion.com>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
	<495A22E0.3040904@networkoblivion.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C670233BC664C@mse17be1.mse17.exchange.ms>

>>  This is
>> because the ua sends it's registration refresh unauthenticated.  The
>> registrar will then push back an authentication challenge request so the
>> ua can prove its identity, at which point the ua then repeats the same
>> transaction, but with authentication credentials attached.
>
>Why does it do that?  Every time I do a debug, I see the first request
>denied as unauthorized and then it always comes right back and gets

Welcome to HTTP Digest authentication. The request has to get challenged to get a new nonce from the server (so as to mitigate replay attacks).

You could TLS and auth off of the client cert, except few devices support that, and you'd have the "overhead" of TCP (which is like bad or something).

-Michael


From jmesquita at gmail.com  Mon Dec 29 20:29:53 2008
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 30 Dec 2008 02:29:53 -0200
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <20081230040049.GA2409@jdc.jasonjgw.net>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
	<20081230013938.184760@gmx.net>
	<20081230021942.GA31689@jdc.jasonjgw.net>
	<20081230040049.GA2409@jdc.jasonjgw.net>
Message-ID: <2A9FC6FF-41F9-47B7-8F5D-EB4FA9A0BCD7@gmail.com>

Thank you Jason, I was just going thru the code when I got your email.  
Saved me up some time. ;)

Mesquita

On Dec 30, 2008, at 2:00 AM, Jason White wrote:

> By the way, the command to exit fs_cli is /exit (or /bye or /quit).
>
> Commands starting with / are handled internally by the  
> process_command()
> function of the CLI, instead of being treated as FreeSWITCH API  
> commands.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Tue Dec 30 06:21:54 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 09:21:54 -0500
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
	<2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>
	<d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>
Message-ID: <965B5A92-4182-489B-8AA0-6E6173C1931B@jerris.com>

Do you realize that calling a busy number and detecting a busy tone  
are COMPLETELY different things, your calling on sip, most likely you  
will not ever get a busy tone to detect but a sip response code when  
you try to call.

Mike

On Dec 30, 2008, at 1:45 AM, Baskar wrote:

> Hi Michael Jerris,
>
> I will explain what i am currently doing :	
> I don't understand
>
> Step 1: From the xlite phone I have dialed a number and we were on  
> the conversation with one extension (1007 is my extension and my  
> mobile No 9841799874)
> Step 2: From the freeswitch console I am executing a javascript file  
> with tone detect like the one below,
>
> My JavaScript :
>
> session1 = new Session();
>
> session1.originate(session1,"{ignore_early_media=false}sofia/internal/1003 at 172.20.201.67 
> ");
>
> session1.execute("tone_detect","busy 480,620 r +30 transfer '1000'  
> 3");
>
> session1.execute("bridge", "sofia/default/9841799874 at 172.20.191.228");
>
> session1.execute("transfer", "9841799874");
>
> session1.hangup;
>             In the above script and in step1 the telephone numbers  
> are same; since the script is not detecting that the phone number is  
> busy.
>
> Hi Michael S Collins,
>
>             Please let me know the script, so that it would be  
> helpful for me.
>
> Thanks for the Reply,
>
>
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Dec 30 06:28:05 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 09:28:05 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A124C.3040006@laposte.net>
References: <495A124C.3040006@laposte.net>
Message-ID: <890434A9-E8B2-441F-9E88-F52B91056CA3@jerris.com>

What revision of FreeSWITCH are you trying with?  I would try with  
current trunk, I have a suspicion we fixed the main issue your running  
into.

Mike

On Dec 30, 2008, at 7:21 AM, rod wrote:

> Hi all,
>
> I know that freeswitch has not been designed as a pure sip
> proxy/registrar, but I'm wondering how many subscribers could be  
> handled
> by FS.
>
> I setup the following test environment:
>    - Kamailio 1.4.2 as the registrar
>    - all invite requests are flowing through FS, even for a call
> between 2 registered subscribers. Many reasons for this: the calls CDR
> are centralized in the same format, I can easily add a billing ID to a
> call, proceed to recording, set the caller as anonymous if  
> requested...
>    - FS is used also as a SBC
>
> There is still a lot of work to do, mainly on the call forwarding
> feature and this is why I'm wondering (simply out of curiosity) what
> could have been achieved using only FS (easier to setup when only one
> equipment is involved :) ).
>
> I'd like to register 40 000 subscribers (if each user registers every
> 60s, you have approx 670 registration per second, this setup is  
> working
> on Kamailio).
>
> I did the following to increase FS performance regarding registration:
>    - put the directory containing users in a RAMDISK
>    - put the db directory in a RAMDISK
>
> with this I was able to reach 190 registration per second (50 without
> the ramdisk) but for one SIP account, not too useful :p (for your
> information I see a huge improvement when switching from 1.0.1  
> phoenix:
> 150cps to FS svn 105xx: 190)
> When trying with 25000 SIP accounts, I got no more than 30cps.
>
> Then I tried to use the odbc mysql for registration, using this I was
> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these
> tests, the presence support has been disabled.
>
> As the IO performance seems to be a bottleneck, I'd like to know if
> there is a way to store the registration in memory only without  
> database
> persistency.
>
> This thread is there only to share tips, not to complain about FS poor
> performance as a SIP registrar when compared to Kamailio. If I compare
> FS to a commercial SBC I'm using in production, I have to say that  
> FS is
> really a great piece of software (lacks only statistics module, snmp,
> and heartbeat redundancy for failover).
>
> regards,
> rod
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kawarod at laposte.net  Tue Dec 30 06:59:58 2008
From: kawarod at laposte.net (rod)
Date: Tue, 30 Dec 2008 18:59:58 +0400
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <890434A9-E8B2-441F-9E88-F52B91056CA3@jerris.com>
References: <495A124C.3040006@laposte.net>
	<890434A9-E8B2-441F-9E88-F52B91056CA3@jerris.com>
Message-ID: <495A376E.2060108@laposte.net>

Hi,

I upgraded today to 10999 with same results.

rod.

Michael Jerris wrote:
> What revision of FreeSWITCH are you trying with?  I would try with  
> current trunk, I have a suspicion we fixed the main issue your running  
> into.
>
> Mike
>
> On Dec 30, 2008, at 7:21 AM, rod wrote:
>
>   
>> Hi all,
>>
>> I know that freeswitch has not been designed as a pure sip
>> proxy/registrar, but I'm wondering how many subscribers could be  
>> handled
>> by FS.
>>
>> I setup the following test environment:
>>    - Kamailio 1.4.2 as the registrar
>>    - all invite requests are flowing through FS, even for a call
>> between 2 registered subscribers. Many reasons for this: the calls CDR
>> are centralized in the same format, I can easily add a billing ID to a
>> call, proceed to recording, set the caller as anonymous if  
>> requested...
>>    - FS is used also as a SBC
>>
>> There is still a lot of work to do, mainly on the call forwarding
>> feature and this is why I'm wondering (simply out of curiosity) what
>> could have been achieved using only FS (easier to setup when only one
>> equipment is involved :) ).
>>
>> I'd like to register 40 000 subscribers (if each user registers every
>> 60s, you have approx 670 registration per second, this setup is  
>> working
>> on Kamailio).
>>
>> I did the following to increase FS performance regarding registration:
>>    - put the directory containing users in a RAMDISK
>>    - put the db directory in a RAMDISK
>>
>> with this I was able to reach 190 registration per second (50 without
>> the ramdisk) but for one SIP account, not too useful :p (for your
>> information I see a huge improvement when switching from 1.0.1  
>> phoenix:
>> 150cps to FS svn 105xx: 190)
>> When trying with 25000 SIP accounts, I got no more than 30cps.
>>
>> Then I tried to use the odbc mysql for registration, using this I was
>> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these
>> tests, the presence support has been disabled.
>>
>> As the IO performance seems to be a bottleneck, I'd like to know if
>> there is a way to store the registration in memory only without  
>> database
>> persistency.
>>
>> This thread is there only to share tips, not to complain about FS poor
>> performance as a SIP registrar when compared to Kamailio. If I compare
>> FS to a commercial SBC I'm using in production, I have to say that  
>> FS is
>> really a great piece of software (lacks only statistics module, snmp,
>> and heartbeat redundancy for failover).
>>
>> regards,
>> rod
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   


From frank at impactfax.com  Tue Dec 30 07:12:28 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 30 Dec 2008 10:12:28 -0500
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <87f2f3b90812291155h3104de95if9d78344aae933f8@mail.gmail.com>
Message-ID: <006c01c96a91$077d3ca0$33014c0a@ws4>

The two endpoints are sip (asterisk) and ulaw.
Thanks.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Monday, December 29, 2008 2:56 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session_record post-processing
 
Curious: what are your endpoints? Also, what codec(s), etc. are you
using? I'm using PCMU with openzap endpoints and I don't get anything
like this at all. I'd like to try and emulate what you've got more
closely to see if I can reproduce the symptoms.

Thanks,
MC
 
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From mike at jerris.com  Tue Dec 30 07:41:47 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 10:41:47 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A376E.2060108@laposte.net>
References: <495A124C.3040006@laposte.net>
	<890434A9-E8B2-441F-9E88-F52B91056CA3@jerris.com>
	<495A376E.2060108@laposte.net>
Message-ID: <9B4CF3F9-D10F-4270-BD2E-111D59199901@jerris.com>

If your not using sqlite, make sure to create indexes on the tables  
created, you should be able to grep the ones we do in sqlite out of  
the code.

Mike

On Dec 30, 2008, at 9:59 AM, rod wrote:

> Hi,
>
> I upgraded today to 10999 with same results.
>
> rod.
>
> Michael Jerris wrote:
>> What revision of FreeSWITCH are you trying with?  I would try with
>> current trunk, I have a suspicion we fixed the main issue your  
>> running
>> into.
>>
>> Mike
>>
>> On Dec 30, 2008, at 7:21 AM, rod wrote:
>>
>>
>>> Hi all,
>>>
>>> I know that freeswitch has not been designed as a pure sip
>>> proxy/registrar, but I'm wondering how many subscribers could be
>>> handled
>>> by FS.
>>>
>>> I setup the following test environment:
>>>   - Kamailio 1.4.2 as the registrar
>>>   - all invite requests are flowing through FS, even for a call
>>> between 2 registered subscribers. Many reasons for this: the calls  
>>> CDR
>>> are centralized in the same format, I can easily add a billing ID  
>>> to a
>>> call, proceed to recording, set the caller as anonymous if
>>> requested...
>>>   - FS is used also as a SBC
>>>
>>> There is still a lot of work to do, mainly on the call forwarding
>>> feature and this is why I'm wondering (simply out of curiosity) what
>>> could have been achieved using only FS (easier to setup when only  
>>> one
>>> equipment is involved :) ).
>>>
>>> I'd like to register 40 000 subscribers (if each user registers  
>>> every
>>> 60s, you have approx 670 registration per second, this setup is
>>> working
>>> on Kamailio).
>>>
>>> I did the following to increase FS performance regarding  
>>> registration:
>>>   - put the directory containing users in a RAMDISK
>>>   - put the db directory in a RAMDISK
>>>
>>> with this I was able to reach 190 registration per second (50  
>>> without
>>> the ramdisk) but for one SIP account, not too useful :p (for your
>>> information I see a huge improvement when switching from 1.0.1
>>> phoenix:
>>> 150cps to FS svn 105xx: 190)
>>> When trying with 25000 SIP accounts, I got no more than 30cps.
>>>
>>> Then I tried to use the odbc mysql for registration, using this I  
>>> was
>>> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all  
>>> these
>>> tests, the presence support has been disabled.
>>>
>>> As the IO performance seems to be a bottleneck, I'd like to know if
>>> there is a way to store the registration in memory only without
>>> database
>>> persistency.
>>>
>>> This thread is there only to share tips, not to complain about FS  
>>> poor
>>> performance as a SIP registrar when compared to Kamailio. If I  
>>> compare
>>> FS to a commercial SBC I'm using in production, I have to say that
>>> FS is
>>> really a great piece of software (lacks only statistics module,  
>>> snmp,
>>> and heartbeat redundancy for failover).
>>>
>>> regards,
>>> rod
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Tue Dec 30 07:46:10 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 10:46:10 -0500
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <006c01c96a91$077d3ca0$33014c0a@ws4>
References: <006c01c96a91$077d3ca0$33014c0a@ws4>
Message-ID: <F196DCED-FCD3-410D-8FCF-94ABF2EF30FB@jerris.com>

Try svn revision 11002 or later, it should now remove the media bugs  
earlier on hangup state before the api_hangup_hook is run.

Mike

On Dec 30, 2008, at 10:12 AM, Frank @ Impact wrote:

> The two endpoints are sip (asterisk) and ulaw.
> Thanks.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Monday, December 29, 2008 2:56 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] session_record post-processing
>
> Curious: what are your endpoints? Also, what codec(s), etc. are you  
> using? I'm using PCMU with openzap endpoints and I don't get  
> anything like this at all. I'd like to try and emulate what you've  
> got more closely to see if I can reproduce the symptoms.

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From e.schmidbauer at gmail.com  Tue Dec 30 09:56:14 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 12:56:14 -0500
Subject: [Freeswitch-users] how to use celt codec
Message-ID: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>

hi, great job with the celt module. im eager to use it but im not sure how.
are there any sip clients that use the celt codec? or is there some other
way to use the celt codec to play audio in a conference? thanks.
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From brian at freeswitch.org  Tue Dec 30 10:06:59 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:06:59 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
Message-ID: <816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>

Well I used mod_portaudio on my Mac with mod_celt to my FreeSWITCH box  
on linux when we developed the module.    Works great!

/b

On Dec 30, 2008, at 11:56 AM, e schmidbauer wrote:

> hi, great job with the celt module. im eager to use it but im not  
> sure how. are there any sip clients that use the celt codec? or is  
> there some other way to use the celt codec to play audio in a  
> conference? thanks.



From can_man at gmx.de  Tue Dec 30 10:10:27 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Tue, 30 Dec 2008 19:10:27 +0100
Subject: [Freeswitch-users] voicemail - Can't find user
Message-ID: <20081230181027.267340@gmx.net>

Hello,

I am trying to get voicemail to run through xml curl, but I get the following error:

2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]

In order to setup user 315 I reply the following to the "directory" request of xml curl:

<user id="315" mailbox="315">
<params>
<param name="password" value="1234"/>
<param name="vm-password" value="0000"/>
</params>
<variables>
<variable name="accountcode" value="315"/>
<variable name="user_context" value="default"/>
<variable name="vm_extension" value="315"/>
<variable name="max_calls" value="1"/>
<variable name="fail_over" value="415"/>
<variable name="cringback" value="us-ring"/>
</variables>
</user>


And in order to send the call to voicemail I do:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="RE Dial Plan For FreeSwitch">
<context name="public">
<extension name="test10000">
<condition field="destination_number" expression="^(10000)$">
<action application="voicemail" data="default $${domain} 315"/>
</condition>
</extension>
</context>
</section>
</document>


Do I maybe have to add the user also at another location? 
Also, I read the following on the wiki: "I figured out that you can respond to both of these requests as follows. Probably the second one is looking for something different, but so far I just ignore it and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
And I do the same, I respond always with the directory response above. Is there a better practice?

It would be great if someone could point out my error. 

Thank you,
Phil


my voicemail conf looks like this:

<configuration name="voicemail.conf" description="Voicemail">
  <settings>
  </settings>
  <profiles>
    <profile name="default">
      <param name="file-extension" value="wav"/>
      <param name="terminator-key" value="#"/>
      <param name="max-login-attempts" value="3"/>
      <param name="digit-timeout" value="10000"/>
      <param name="min-record-len" value="3"/>
      <param name="max-record-len" value="300"/>
      <param name="tone-spec" value="%(1000, 0, 640)"/>
      <param name="callback-dialplan" value="XML"/>
      <param name="callback-context" value="default"/>
      <param name="play-new-messages-key" value="1"/>
      <param name="play-saved-messages-key" value="2"/>
      <param name="main-menu-key" value="0"/>
      <param name="config-menu-key" value="5"/>
      <param name="record-greeting-key" value="1"/>
      <param name="choose-greeting-key" value="2"/>
      <param name="change-pass-key" value="6"/>
      <param name="record-name-key" value="3"/>
      <param name="record-file-key" value="3"/>
      <param name="listen-file-key" value="1"/>
      <param name="save-file-key" value="2"/>
      <param name="delete-file-key" value="7"/>
      <param name="undelete-file-key" value="8"/>
      <param name="email-key" value="4"/>
      <param name="pause-key" value="0"/>
      <param name="restart-key" value="1"/>
      <param name="ff-key" value="6"/>
      <param name="rew-key" value="4"/>
      <param name="record-silence-threshold" value="200"/>
      <param name="record-silence-hits" value="2"/>
      <param name="web-template-file" value="web-vm.tpl"/>
      <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
      <!--<param name="record-sample-rate" value="11025"/>-->
      <!-- the next two both must be set for this to be enabled
           the extension is in the format of <dest> [<dialplan>] [<context>]
       -->
      <param name="operator-extension" value="operator XML default"/>
      <param name="operator-key" value="9"/>
      <param name="vmain-extension" value="vmain XML default"/>
      <param name="vmain-key" value="*"/>
      <!-- playback created files as soon as they were recorded by default -->
      <!--<param name="auto-playback-recordings" value="true"/>-->
      <email>
        <param name="template-file" value="voicemail.tpl"/>
        <param name="notify-template-file" value="notify-voicemail.tpl"/>
        <!-- this is the format voicemail_time will have -->
        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
        <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
      </email>
      <!--<param name="storage-dir" value="/tmp"/>-->
      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
      <!--<param name="record-comment" value="Your Comment"/>-->
      <!--<param name="record-title" value="Your Title"/>-->
      <!--<param name="record-copyright" value="Your Copyright"/>-->
    </profile>
  </profiles>
</configuration>






the debug output:


2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message() Asked to send early media by sofia/external/anonymous at sipgate.de
2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/anonymous at sipgate.de] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms: 20
2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms
2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message() Ring SDP:
v=0
o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
s=FreeSWITCH
c=IN IP4 89.49.116.108
t=0 0
m=audio 61125 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de!
2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de!
2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/anonymous at sipgate.de entering state [early]


2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]


2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117 switch_ivr_phrase_macro() No language specified - Using [en]
2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269 switch_ivr_phrase_macro() Handle play-file:[voicemail/vm-goodbye.wav] (en:en)
2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/external/anonymous at sipgate.de receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file
2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de [KILL]
2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE going to sleep
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_HANGUP
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP
2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous at sipgate.de hanging up, cause: NORMAL_CLEARING
2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 480
2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP going to sleep
2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Locked, Waiting on external entities
2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Ended
2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]
-- 
Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL 
f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a


From brian at freeswitch.org  Tue Dec 30 10:19:16 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:19:16 -0600
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <20081230181027.267340@gmx.net>
References: <20081230181027.267340@gmx.net>
Message-ID: <268644AF-1EC2-4645-9473-7E588BF25547@freeswitch.org>

what svn rev are you on?

/b

On Dec 30, 2008, at 12:10 PM, can_man at gmx.de wrote:

> <user id="315" mailbox="315">
> <params>
> <param name="password" value="1234"/>
> <param name="vm-password" value="0000"/>
> </params>
> <variables>
> <variable name="accountcode" value="315"/>
> <variable name="user_context" value="default"/>
> <variable name="vm_extension" value="315"/>
> <variable name="max_calls" value="1"/>
> <variable name="fail_over" value="415"/>
> <variable name="cringback" value="us-ring"/>
> </variables>
> </user>



From intralanman at freeswitch.org  Tue Dec 30 10:26:36 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Tue, 30 Dec 2008 18:26:36 +0000
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <20081230181027.267340@gmx.net>
References: <20081230181027.267340@gmx.net>
Message-ID: <495A67DC.9060203@freeswitch.org>

you need to add something similar to the following to your directory 
request:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="arbitrary stuff here">


-Ray




can_man at gmx.de wrote:
> Hello,
>
> I am trying to get voicemail to run through xml curl, but I get the following error:
>
> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]
>
> In order to setup user 315 I reply the following to the "directory" request of xml curl:
>
> <user id="315" mailbox="315">
> <params>
> <param name="password" value="1234"/>
> <param name="vm-password" value="0000"/>
> </params>
> <variables>
> <variable name="accountcode" value="315"/>
> <variable name="user_context" value="default"/>
> <variable name="vm_extension" value="315"/>
> <variable name="max_calls" value="1"/>
> <variable name="fail_over" value="415"/>
> <variable name="cringback" value="us-ring"/>
> </variables>
> </user>
>
>
> And in order to send the call to voicemail I do:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="RE Dial Plan For FreeSwitch">
> <context name="public">
> <extension name="test10000">
> <condition field="destination_number" expression="^(10000)$">
> <action application="voicemail" data="default $${domain} 315"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
>
> Do I maybe have to add the user also at another location? 
> Also, I read the following on the wiki: "I figured out that you can respond to both of these requests as follows. Probably the second one is looking for something different, but so far I just ignore it and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
> And I do the same, I respond always with the directory response above. Is there a better practice?
>
> It would be great if someone could point out my error. 
>
> Thank you,
> Phil
>
>
> my voicemail conf looks like this:
>
> <configuration name="voicemail.conf" description="Voicemail">
>   <settings>
>   </settings>
>   <profiles>
>     <profile name="default">
>       <param name="file-extension" value="wav"/>
>       <param name="terminator-key" value="#"/>
>       <param name="max-login-attempts" value="3"/>
>       <param name="digit-timeout" value="10000"/>
>       <param name="min-record-len" value="3"/>
>       <param name="max-record-len" value="300"/>
>       <param name="tone-spec" value="%(1000, 0, 640)"/>
>       <param name="callback-dialplan" value="XML"/>
>       <param name="callback-context" value="default"/>
>       <param name="play-new-messages-key" value="1"/>
>       <param name="play-saved-messages-key" value="2"/>
>       <param name="main-menu-key" value="0"/>
>       <param name="config-menu-key" value="5"/>
>       <param name="record-greeting-key" value="1"/>
>       <param name="choose-greeting-key" value="2"/>
>       <param name="change-pass-key" value="6"/>
>       <param name="record-name-key" value="3"/>
>       <param name="record-file-key" value="3"/>
>       <param name="listen-file-key" value="1"/>
>       <param name="save-file-key" value="2"/>
>       <param name="delete-file-key" value="7"/>
>       <param name="undelete-file-key" value="8"/>
>       <param name="email-key" value="4"/>
>       <param name="pause-key" value="0"/>
>       <param name="restart-key" value="1"/>
>       <param name="ff-key" value="6"/>
>       <param name="rew-key" value="4"/>
>       <param name="record-silence-threshold" value="200"/>
>       <param name="record-silence-hits" value="2"/>
>       <param name="web-template-file" value="web-vm.tpl"/>
>       <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
>       <!--<param name="record-sample-rate" value="11025"/>-->
>       <!-- the next two both must be set for this to be enabled
>            the extension is in the format of <dest> [<dialplan>] [<context>]
>        -->
>       <param name="operator-extension" value="operator XML default"/>
>       <param name="operator-key" value="9"/>
>       <param name="vmain-extension" value="vmain XML default"/>
>       <param name="vmain-key" value="*"/>
>       <!-- playback created files as soon as they were recorded by default -->
>       <!--<param name="auto-playback-recordings" value="true"/>-->
>       <email>
>         <param name="template-file" value="voicemail.tpl"/>
>         <param name="notify-template-file" value="notify-voicemail.tpl"/>
>         <!-- this is the format voicemail_time will have -->
>         <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>         <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
>       </email>
>       <!--<param name="storage-dir" value="/tmp"/>-->
>       <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
>       <!--<param name="record-comment" value="Your Comment"/>-->
>       <!--<param name="record-title" value="Your Title"/>-->
>       <!--<param name="record-copyright" value="Your Copyright"/>-->
>     </profile>
>   </profiles>
> </configuration>
>
>
>
>
>
>
> the debug output:
>
>
> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message() Asked to send early media by sofia/external/anonymous at sipgate.de
> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/anonymous at sipgate.de] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms: 20
> 2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms
> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message() Ring SDP:
> v=0
> o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
> s=FreeSWITCH
> c=IN IP4 89.49.116.108
> t=0 0
> m=audio 61125 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de!
> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de!
> 2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de [BREAK]
> 2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/anonymous at sipgate.de entering state [early]
>
>
> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]
>
>
> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117 switch_ivr_phrase_macro() No language specified - Using [en]
> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269 switch_ivr_phrase_macro() Handle play-file:[voicemail/vm-goodbye.wav] (en:en)
> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
> 2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/external/anonymous at sipgate.de receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
> 2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file
> 2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de [CS_EXECUTE] [NORMAL_CLEARING]
> 2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de [KILL]
> 2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE going to sleep
> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_HANGUP
> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP
> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous at sipgate.de hanging up, cause: NORMAL_CLEARING
> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 480
> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP going to sleep
> 2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Locked, Waiting on external entities
> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Ended
> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]
>   



From brian at freeswitch.org  Tue Dec 30 10:31:45 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:31:45 -0600
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <495A67DC.9060203@freeswitch.org>
References: <20081230181027.267340@gmx.net> <495A67DC.9060203@freeswitch.org>
Message-ID: <46AB3E92-2A2C-4DF9-93F0-13D7CC314BA9@freeswitch.org>

and end that with

</section>
</document>

:P

On Dec 30, 2008, at 12:26 PM, Raymond Chandler wrote:

> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="arbitrary stuff here">



From e.schmidbauer at gmail.com  Tue Dec 30 10:33:01 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 13:33:01 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
Message-ID: <2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>

Could you explain in a more detail how you set that up?

On Tue, Dec 30, 2008 at 1:06 PM, Brian West <brian at freeswitch.org> wrote:

> Well I used mod_portaudio on my Mac with mod_celt to my FreeSWITCH box
> on linux when we developed the module.    Works great!
>
> /b
>
> On Dec 30, 2008, at 11:56 AM, e schmidbauer wrote:
>
> > hi, great job with the celt module. im eager to use it but im not
> > sure how. are there any sip clients that use the celt codec? or is
> > there some other way to use the celt codec to play audio in a
> > conference? thanks.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Dec 30 10:36:44 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:36:44 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
Message-ID: <B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>

http://wiki.freeswitch.org/wiki/Freeswitch_softphone

/b

On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:

> Could you explain in a more detail how you set that up?



From e.schmidbauer at gmail.com  Tue Dec 30 10:46:38 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 13:46:38 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
Message-ID: <2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>

i have port audio setup but when i do a 'pa call <extension>' it enters the
conference using the L16 codec. is there a way to use celt codec instead of
the L16?

On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian at freeswitch.org> wrote:

> http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>
> /b
>
> On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
>
> > Could you explain in a more detail how you set that up?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Dec 30 10:47:24 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:47:24 -0600
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <495A67DC.9060203@freeswitch.org>
References: <20081230181027.267340@gmx.net> <495A67DC.9060203@freeswitch.org>
Message-ID: <AF0F57E5-17FA-4CB5-8F76-D93E13EFBF28@freeswitch.org>

I would update to the new method using groups

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="arbitrary stuff here">
	<domain name="foo.com">
		<groups>
       		<group name="default">
         		<users>
          			<user id="315" mailbox="315">
						<params>
							<param name="password" value="1234"/>
							<param name="vm-password" value="0000"/>
						</params>
						<variables>
							<variable name="accountcode" value="315"/>
							<variable name="user_context" value="default"/>
							<variable name="vm_extension" value="315"/>
							<variable name="max_calls" value="1"/>
							<variable name="fail_over" value="415"/>
							<variable name="cringback" value="us-ring"/>
						</variables>
					</user>

         		</users>
       		</group>
		</groups>
	</domain>
</section>
</document>


/b


On Dec 30, 2008, at 12:26 PM, Raymond Chandler wrote:

> you need to add something similar to the following to your directory
> request:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="arbitrary stuff here">
>
>
> -Ray
>
>
>
>
> can_man at gmx.de wrote:
>> Hello,
>>
>> I am trying to get voicemail to run through xml curl, but I get the  
>> following error:
>>
>> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737  
>> voicemail_leave_main() Can't find user [315 at 192.168.178.22]
>>
>> In order to setup user 315 I reply the following to the "directory"  
>> request of xml curl:
>>
>> <user id="315" mailbox="315">
>> <params>
>> <param name="password" value="1234"/>
>> <param name="vm-password" value="0000"/>
>> </params>
>> <variables>
>> <variable name="accountcode" value="315"/>
>> <variable name="user_context" value="default"/>
>> <variable name="vm_extension" value="315"/>
>> <variable name="max_calls" value="1"/>
>> <variable name="fail_over" value="415"/>
>> <variable name="cringback" value="us-ring"/>
>> </variables>
>> </user>
>>
>>
>> And in order to send the call to voicemail I do:
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>> <context name="public">
>> <extension name="test10000">
>> <condition field="destination_number" expression="^(10000)$">
>> <action application="voicemail" data="default $${domain} 315"/>
>> </condition>
>> </extension>
>> </context>
>> </section>
>> </document>
>>
>>
>> Do I maybe have to add the user also at another location?
>> Also, I read the following on the wiki: "I figured out that you can  
>> respond to both of these requests as follows. Probably the second  
>> one is looking for something different, but so far I just ignore it  
>> and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
>> And I do the same, I respond always with the directory response  
>> above. Is there a better practice?
>>
>> It would be great if someone could point out my error.
>>
>> Thank you,
>> Phil
>>
>>
>> my voicemail conf looks like this:
>>
>> <configuration name="voicemail.conf" description="Voicemail">
>>  <settings>
>>  </settings>
>>  <profiles>
>>    <profile name="default">
>>      <param name="file-extension" value="wav"/>
>>      <param name="terminator-key" value="#"/>
>>      <param name="max-login-attempts" value="3"/>
>>      <param name="digit-timeout" value="10000"/>
>>      <param name="min-record-len" value="3"/>
>>      <param name="max-record-len" value="300"/>
>>      <param name="tone-spec" value="%(1000, 0, 640)"/>
>>      <param name="callback-dialplan" value="XML"/>
>>      <param name="callback-context" value="default"/>
>>      <param name="play-new-messages-key" value="1"/>
>>      <param name="play-saved-messages-key" value="2"/>
>>      <param name="main-menu-key" value="0"/>
>>      <param name="config-menu-key" value="5"/>
>>      <param name="record-greeting-key" value="1"/>
>>      <param name="choose-greeting-key" value="2"/>
>>      <param name="change-pass-key" value="6"/>
>>      <param name="record-name-key" value="3"/>
>>      <param name="record-file-key" value="3"/>
>>      <param name="listen-file-key" value="1"/>
>>      <param name="save-file-key" value="2"/>
>>      <param name="delete-file-key" value="7"/>
>>      <param name="undelete-file-key" value="8"/>
>>      <param name="email-key" value="4"/>
>>      <param name="pause-key" value="0"/>
>>      <param name="restart-key" value="1"/>
>>      <param name="ff-key" value="6"/>
>>      <param name="rew-key" value="4"/>
>>      <param name="record-silence-threshold" value="200"/>
>>      <param name="record-silence-hits" value="2"/>
>>      <param name="web-template-file" value="web-vm.tpl"/>
>>      <!-- if you need to change the sample rate of the recorded  
>> files e.g. gmail voicemail player -->
>>      <!--<param name="record-sample-rate" value="11025"/>-->
>>      <!-- the next two both must be set for this to be enabled
>>           the extension is in the format of <dest> [<dialplan>]  
>> [<context>]
>>       -->
>>      <param name="operator-extension" value="operator XML default"/>
>>      <param name="operator-key" value="9"/>
>>      <param name="vmain-extension" value="vmain XML default"/>
>>      <param name="vmain-key" value="*"/>
>>      <!-- playback created files as soon as they were recorded by  
>> default -->
>>      <!--<param name="auto-playback-recordings" value="true"/>-->
>>      <email>
>>        <param name="template-file" value="voicemail.tpl"/>
>>        <param name="notify-template-file" value="notify- 
>> voicemail.tpl"/>
>>        <!-- this is the format voicemail_time will have -->
>>        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>>        <param name="email-from" value="${voicemail_account}@$ 
>> {voicemail_domain}"/>
>>      </email>
>>      <!--<param name="storage-dir" value="/tmp"/>-->
>>      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
>>      <!--<param name="record-comment" value="Your Comment"/>-->
>>      <!--<param name="record-title" value="Your Title"/>-->
>>      <!--<param name="record-copyright" value="Your Copyright"/>-->
>>    </profile>
>>  </profiles>
>> </configuration>
>>
>>
>>
>>
>>
>>
>> the debug output:
>>
>>
>> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message()  
>> Asked to send early media by sofia/external/anonymous at sipgate.de
>> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497  
>> sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
>> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825  
>> sofia_glue_activate_rtp() AUDIO RTP [sofia/external/anonymous at sipgate.de 
>> ] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms:  
>> 20
>> 2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create()  
>> Starting timer [soft] 160 bytes per 20000ms
>> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message()  
>> Ring SDP:
>> v=0
>> o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
>> s=FreeSWITCH
>> c=IN IP4 89.49.116.108
>> t=0 0
>> m=audio 61125 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316  
>> sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de 
>> !
>> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316  
>> sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de 
>> !
>> 2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510  
>> switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de 
>>  [BREAK]
>> 2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state()  
>> Channel sofia/external/anonymous at sipgate.de entering state [early]
>>
>>
>> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737  
>> voicemail_leave_main() Can't find user [315 at 192.168.178.22]
>>
>>
>> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117  
>> switch_ivr_phrase_macro() No language specified - Using [en]
>> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269  
>> switch_ivr_phrase_macro() Handle play-file:[voicemail/vm- 
>> goodbye.wav] (en:en)
>> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932  
>> switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
>> 2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655  
>> switch_core_session_write_frame() sofia/external/ 
>> anonymous at sipgate.de receive message  
>> [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
>> 2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222  
>> switch_ivr_play_file() done playing file
>> 2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168  
>> switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de 
>>  [CS_EXECUTE] [NORMAL_CLEARING]
>> 2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494  
>> switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de 
>>  [KILL]
>> 2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806  
>> switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de 
>>  [BREAK]
>> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442  
>> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
>> State EXECUTE going to sleep
>> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369  
>> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
>> Running State Change CS_HANGUP
>> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400  
>> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
>> State HANGUP
>> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup()  
>> Channel sofia/external/anonymous at sipgate.de hanging up, cause:  
>> NORMAL_CLEARING
>> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup()  
>> Responding to INVITE with: 480
>> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46  
>> switch_core_standard_on_hangup() sofia/external/ 
>> anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
>> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400  
>> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
>> State HANGUP going to sleep
>> 2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938  
>> switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de 
>> ) Locked, Waiting on external entities
>> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956  
>> switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de 
>> ) Ended
>> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958  
>> switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de 
>>  [CS_HANGUP]
>>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Dec 30 10:55:39 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:55:39 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
Message-ID: <05DFBEBE-01EE-4AA9-BC5D-B4C5403B4368@freeswitch.org>

You need to use CELT between FS and another FS box, L16 is from the PA  
to the Conference no need to encode it to celt and then decode it  
again.. it never hits the wire.

/b

On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:

> i have port audio setup but when i do a 'pa call <extension>' it  
> enters the conference using the L16 codec. is there a way to use  
> celt codec instead of the L16?



From brian at freeswitch.org  Tue Dec 30 11:06:10 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 13:06:10 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
Message-ID: <A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>

OK here try this..

in portaudio.conf.xml

     <param name="sample-rate" value="48000"/>
     <param name="codec-ms" value="10"/>


in dialplan/default.xml

     <extension name="sip_uri">
       <condition field="destination_number" expression="^sip:(.*)$">

         <action application="bridge"  
data="{absolute_codec_string=CELT at 48000h@10i}sofia/${use_profile}/sip: 
$1"/>
       </condition>
     </extension>

save that
then

pa call sip:886 at taz.bkw.org:5080

/b


On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:

> i have port audio setup but when i do a 'pa call <extension>' it  
> enters the conference using the L16 codec. is there a way to use  
> celt codec instead of the L16?
>
> On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian at freeswitch.org>  
> wrote:
> http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>
> /b
>
> On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
>
> > Could you explain in a more detail how you set that up?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From can_man at gmx.de  Tue Dec 30 11:08:01 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Tue, 30 Dec 2008 20:08:01 +0100
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <AF0F57E5-17FA-4CB5-8F76-D93E13EFBF28@freeswitch.org>
References: <20081230181027.267340@gmx.net> <495A67DC.9060203@freeswitch.org>
	<AF0F57E5-17FA-4CB5-8F76-D93E13EFBF28@freeswitch.org>
Message-ID: <20081230190801.302740@gmx.net>

Hello,

thank you for your answers. I have added the start and end tags to my xml, but nothing has changed.
However, the "groups XML" did work with my server's IP as: <domain name="192.168.178.22"> - thank you Brian.

If someone can shade some light into this quote from the wiki:  "I figured out that you can respond to both of these requests as follows. Probably the second one is looking for something different, but so far I just ignore it and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22

I will re-write the whole "directory" section on the wiki. For now I will add the "group" reply.

Thank you,
Phil

Ps: if it is still of interest, my svn version is:

URL: http://svn.freeswitch.org/svn/freeswitch/trunk
Repository Root: http://svn.freeswitch.org/svn
Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2
Revision: 10988
Node Kind: directory
Schedule: normal
Last Changed Author: brian
Last Changed Rev: 10983
Last Changed Date: 2008-12-29 06:27:53 +0100 (Mon, 29 Dec 2008)




> I would update to the new method using groups
> 
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="arbitrary stuff here">
> 	<domain name="foo.com">
> 		<groups>
>        		<group name="default">
>          		<users>
>           			<user id="315" mailbox="315">
> 						<params>
> 							<param name="password" value="1234"/>
> 							<param name="vm-password" value="0000"/>
> 						</params>
> 						<variables>
> 							<variable name="accountcode" value="315"/>
> 							<variable name="user_context" value="default"/>
> 							<variable name="vm_extension" value="315"/>
> 							<variable name="max_calls" value="1"/>
> 							<variable name="fail_over" value="415"/>
> 							<variable name="cringback" value="us-ring"/>
> 						</variables>
> 					</user>
> 
>          		</users>
>        		</group>
> 		</groups>
> 	</domain>
> </section>
> </document>
> 
> 
> /b
> 
> 
> On Dec 30, 2008, at 12:26 PM, Raymond Chandler wrote:
> 
> > you need to add something similar to the following to your directory
> > request:
> >
> > <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> > <document type="freeswitch/xml">
> > <section name="directory" description="arbitrary stuff here">
> >
> >
> > -Ray
> >
> >
> >
> >
> > can_man at gmx.de wrote:
> >> Hello,
> >>
> >> I am trying to get voicemail to run through xml curl, but I get the  
> >> following error:
> >>
> >> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737  
> >> voicemail_leave_main() Can't find user [315 at 192.168.178.22]
> >>
> >> In order to setup user 315 I reply the following to the "directory"  
> >> request of xml curl:
> >>
> >> <user id="315" mailbox="315">
> >> <params>
> >> <param name="password" value="1234"/>
> >> <param name="vm-password" value="0000"/>
> >> </params>
> >> <variables>
> >> <variable name="accountcode" value="315"/>
> >> <variable name="user_context" value="default"/>
> >> <variable name="vm_extension" value="315"/>
> >> <variable name="max_calls" value="1"/>
> >> <variable name="fail_over" value="415"/>
> >> <variable name="cringback" value="us-ring"/>
> >> </variables>
> >> </user>
> >>
> >>
> >> And in order to send the call to voicemail I do:
> >>
> >> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> >> <document type="freeswitch/xml">
> >> <section name="dialplan" description="RE Dial Plan For FreeSwitch">
> >> <context name="public">
> >> <extension name="test10000">
> >> <condition field="destination_number" expression="^(10000)$">
> >> <action application="voicemail" data="default $${domain} 315"/>
> >> </condition>
> >> </extension>
> >> </context>
> >> </section>
> >> </document>
> >>
> >>
> >> Do I maybe have to add the user also at another location?
> >> Also, I read the following on the wiki: "I figured out that you can  
> >> respond to both of these requests as follows. Probably the second  
> >> one is looking for something different, but so far I just ignore it  
> >> and throw out the same stuff." at
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
> >> And I do the same, I respond always with the directory response  
> >> above. Is there a better practice?
> >>
> >> It would be great if someone could point out my error.
> >>
> >> Thank you,
> >> Phil
> >>
> >>
> >> my voicemail conf looks like this:
> >>
> >> <configuration name="voicemail.conf" description="Voicemail">
> >>  <settings>
> >>  </settings>
> >>  <profiles>
> >>    <profile name="default">
> >>      <param name="file-extension" value="wav"/>
> >>      <param name="terminator-key" value="#"/>
> >>      <param name="max-login-attempts" value="3"/>
> >>      <param name="digit-timeout" value="10000"/>
> >>      <param name="min-record-len" value="3"/>
> >>      <param name="max-record-len" value="300"/>
> >>      <param name="tone-spec" value="%(1000, 0, 640)"/>
> >>      <param name="callback-dialplan" value="XML"/>
> >>      <param name="callback-context" value="default"/>
> >>      <param name="play-new-messages-key" value="1"/>
> >>      <param name="play-saved-messages-key" value="2"/>
> >>      <param name="main-menu-key" value="0"/>
> >>      <param name="config-menu-key" value="5"/>
> >>      <param name="record-greeting-key" value="1"/>
> >>      <param name="choose-greeting-key" value="2"/>
> >>      <param name="change-pass-key" value="6"/>
> >>      <param name="record-name-key" value="3"/>
> >>      <param name="record-file-key" value="3"/>
> >>      <param name="listen-file-key" value="1"/>
> >>      <param name="save-file-key" value="2"/>
> >>      <param name="delete-file-key" value="7"/>
> >>      <param name="undelete-file-key" value="8"/>
> >>      <param name="email-key" value="4"/>
> >>      <param name="pause-key" value="0"/>
> >>      <param name="restart-key" value="1"/>
> >>      <param name="ff-key" value="6"/>
> >>      <param name="rew-key" value="4"/>
> >>      <param name="record-silence-threshold" value="200"/>
> >>      <param name="record-silence-hits" value="2"/>
> >>      <param name="web-template-file" value="web-vm.tpl"/>
> >>      <!-- if you need to change the sample rate of the recorded  
> >> files e.g. gmail voicemail player -->
> >>      <!--<param name="record-sample-rate" value="11025"/>-->
> >>      <!-- the next two both must be set for this to be enabled
> >>           the extension is in the format of <dest> [<dialplan>]  
> >> [<context>]
> >>       -->
> >>      <param name="operator-extension" value="operator XML default"/>
> >>      <param name="operator-key" value="9"/>
> >>      <param name="vmain-extension" value="vmain XML default"/>
> >>      <param name="vmain-key" value="*"/>
> >>      <!-- playback created files as soon as they were recorded by  
> >> default -->
> >>      <!--<param name="auto-playback-recordings" value="true"/>-->
> >>      <email>
> >>        <param name="template-file" value="voicemail.tpl"/>
> >>        <param name="notify-template-file" value="notify- 
> >> voicemail.tpl"/>
> >>        <!-- this is the format voicemail_time will have -->
> >>        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
> >>        <param name="email-from" value="${voicemail_account}@$ 
> >> {voicemail_domain}"/>
> >>      </email>
> >>      <!--<param name="storage-dir" value="/tmp"/>-->
> >>      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
> >>      <!--<param name="record-comment" value="Your Comment"/>-->
> >>      <!--<param name="record-title" value="Your Title"/>-->
> >>      <!--<param name="record-copyright" value="Your Copyright"/>-->
> >>    </profile>
> >>  </profiles>
> >> </configuration>
> >>
> >>
> >>
> >>
> >>
> >>
> >> the debug output:
> >>
> >>
> >> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message()  
> >> Asked to send early media by sofia/external/anonymous at sipgate.de
> >> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497  
> >> sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
> >> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825  
> >> sofia_glue_activate_rtp() AUDIO RTP
> [sofia/external/anonymous at sipgate.de 
> >> ] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms:  
> >> 20
> >> 2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create()  
> >> Starting timer [soft] 160 bytes per 20000ms
> >> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message()  
> >> Ring SDP:
> >> v=0
> >> o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
> >> s=FreeSWITCH
> >> c=IN IP4 89.49.116.108
> >> t=0 0
> >> m=audio 61125 RTP/AVP 8 101
> >> a=rtpmap:8 PCMA/8000
> >> a=rtpmap:101 telephone-event/8000
> >> a=fmtp:101 0-16
> >> a=silenceSupp:off - - - -
> >> a=ptime:20
> >> a=sendrecv
> >>
> >> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316  
> >> sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de 
> >> !
> >> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316  
> >> sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de 
> >> !
> >> 2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510  
> >> switch_core_session_perform_receive_message() Send signal
> sofia/external/anonymous at sipgate.de 
> >>  [BREAK]
> >> 2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state()  
> >> Channel sofia/external/anonymous at sipgate.de entering state [early]
> >>
> >>
> >> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737  
> >> voicemail_leave_main() Can't find user [315 at 192.168.178.22]
> >>
> >>
> >> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117  
> >> switch_ivr_phrase_macro() No language specified - Using [en]
> >> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269  
> >> switch_ivr_phrase_macro() Handle play-file:[voicemail/vm- 
> >> goodbye.wav] (en:en)
> >> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932  
> >> switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
> >> 2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655  
> >> switch_core_session_write_frame() sofia/external/ 
> >> anonymous at sipgate.de receive message  
> >> [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
> >> 2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222  
> >> switch_ivr_play_file() done playing file
> >> 2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168  
> >> switch_core_standard_on_execute() Hangup
> sofia/external/anonymous at sipgate.de 
> >>  [CS_EXECUTE] [NORMAL_CLEARING]
> >> 2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494  
> >> switch_channel_perform_hangup() Send signal
> sofia/external/anonymous at sipgate.de 
> >>  [KILL]
> >> 2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806  
> >> switch_core_session_signal_state_change() Send signal
> sofia/external/anonymous at sipgate.de 
> >>  [BREAK]
> >> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442  
> >> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
> >> State EXECUTE going to sleep
> >> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369  
> >> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
> >> Running State Change CS_HANGUP
> >> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400  
> >> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
> >> State HANGUP
> >> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup()  
> >> Channel sofia/external/anonymous at sipgate.de hanging up, cause:  
> >> NORMAL_CLEARING
> >> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup()  
> >> Responding to INVITE with: 480
> >> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46  
> >> switch_core_standard_on_hangup() sofia/external/ 
> >> anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
> >> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400  
> >> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
> >> State HANGUP going to sleep
> >> 2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938  
> >> switch_core_session_thread() Session 2
> (sofia/external/anonymous at sipgate.de 
> >> ) Locked, Waiting on external entities
> >> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956  
> >> switch_core_session_thread() Session 2
> (sofia/external/anonymous at sipgate.de 
> >> ) Ended
> >> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958  
> >> switch_core_session_thread() Close Channel
> sofia/external/anonymous at sipgate.de 
> >>  [CS_HANGUP]
> >>
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger


From brian at freeswitch.org  Tue Dec 30 11:14:23 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 13:14:23 -0600
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <20081230190801.302740@gmx.net>
References: <20081230181027.267340@gmx.net> <495A67DC.9060203@freeswitch.org>
	<AF0F57E5-17FA-4CB5-8F76-D93E13EFBF28@freeswitch.org>
	<20081230190801.302740@gmx.net>
Message-ID: <91A83654-30D9-4DDC-A93F-E9B3109BEF65@freeswitch.org>

Thank you... it needed to be updated ;)

btw we now have a group/ endpoint.. so you can call group/sales at domain  
and ring everyone in the group.  Go check out the default config it  
has use examples.

/b

On Dec 30, 2008, at 1:08 PM, can_man at gmx.de wrote:

> I will re-write the whole "directory" section on the wiki. For now I  
> will add the "group" reply.
>
> Thank you,
> Phil



From e.schmidbauer at gmail.com  Tue Dec 30 11:50:21 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 14:50:21 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
Message-ID: <2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>

i did as your said and got some errors in the console....
after i run pa call sip:886 at ww2.bwrl.org:5080 i get.....
2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
relaxxplayer->sip:886 at ww2.bwrl.org:5080 in context default
2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid
Profile
2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close
Channel N/A [CS_NEW]
2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate()
Cannot create outgoing channel of type [sofia] cause:
[DESTINATION_OUT_OF_ORDER]
2008-12-30 14:45:25 [INFO] mod_dptools.c:1891 audio_bridge_function()
Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER

any suggestions?


On Tue, Dec 30, 2008 at 2:06 PM, Brian West <brian at freeswitch.org> wrote:

> OK here try this..
> in portaudio.conf.xml
>
>     <param name="sample-rate" value="48000"/>
>
>
>     <param name="codec-ms" value="10"/>
>
>
> in dialplan/default.xml
>
>     <extension name="sip_uri">
>
>
>       <condition field="destination_number" expression="^sip:(.*)$">
>
>
>
>
>
>         <action application="bridge"
> data="{absolute_codec_string=CELT at 48000h@10i}sofia/${use_profile}/sip:$1"/>
>
>       </condition>
>
>
>     </extension>
>
> save that
> then
>
> pa call sip:886 at taz.bkw.org:5080
>
> /b
>
>
> On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:
>
> i have port audio setup but when i do a 'pa call <extension>' it enters the
> conference using the L16 codec. is there a way to use celt codec instead of
> the L16?
>
> On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian at freeswitch.org> wrote:
>
>> http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>
>> /b
>>
>> On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
>>
>> > Could you explain in a more detail how you set that up?
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Dec 30 12:01:17 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Dec 2008 12:01:17 -0800
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <B95277C3-C7E9-43A7-B4F4-6B55B7BD17CE@myrvold.org>
References: <20081230040049.GA2409@jdc.jasonjgw.net>
	<C57F2289.4B7B7%krice@suspicious.org>
	<87f2f3b90812292324r1c070458sb20c96db5df2be53@mail.gmail.com>
	<B95277C3-C7E9-43A7-B4F4-6B55B7BD17CE@myrvold.org>
Message-ID: <87f2f3b90812301201h2a0b06ebu3acaf72c15c0817d@mail.gmail.com>

Thanks for the heads up. I was able to confirm this behavior. I changed the
wiki to read "/noevents".
-MC

On Tue, Dec 30, 2008 at 3:03 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:

> I found out that both "/event" and "/events" worked as commands, but only
> "/noevents" worked, not "/noevent", although the Wiki says "/noevent".
> Ivan
>
> Den 30. des.. 2008 kl. 08:24 skrev Michael Collins:
>
> Ken,
>
> Thanks for the clarification. I will make a note of this in the wiki. Also,
> can you hum a few bars and tell us what the "/filter" command does?
>
> -MC
>
> On Mon, Dec 29, 2008 at 10:57 PM, Ken Rice <krice at suspicious.org> wrote:
>
>> You can also use the ... Command to exit fs_cli and there are a few more
>> commands that are locally processed. (note: on the FS main console ...
>> Will
>> cause fs to shutdown. Fs_cli interprets this locally and it does not shut
>> down the main system. You stll need to do fsclt shutdown or something
>> similar)
>>
>> See the wiki for more information M Collins did a pretty good job
>> documenting it
>>
>> K
>>
>>
>> > From: Jason White <jason at jasonjgw.net>
>> > Reply-To: <freeswitch-users at lists.freeswitch.org>
>> > Date: Tue, 30 Dec 2008 15:00:49 +1100
>> > To: <freeswitch-users at lists.freeswitch.org>
>> > Subject: Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
>> > Available!
>> >
>> > By the way, the command to exit fs_cli is /exit (or /bye or /quit).
>> >
>> > Commands starting with / are handled internally by the process_command()
>> > function of the CLI, instead of being treated as FreeSWITCH API
>> commands.
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Dec 30 12:05:49 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 14:05:49 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
Message-ID: <50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>

Yes actually call sip:886 at taz.bkw.org:5080   I set it up specifically  
for you to test :P

/b

On Dec 30, 2008, at 1:50 PM, e schmidbauer wrote:

> i did as your said and got some errors in the console....
> after i run pa call sip:886 at ww2.bwrl.org:5080 i get.....
> 2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()  
> Processing relaxxplayer->sip:886 at ww2.bwrl.org:5080 in context default
> 2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel()  
> Invalid Profile
> 2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540  
> sofia_outgoing_channel() Close Channel N/A [CS_NEW]
> 2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116  
> switch_ivr_originate() Cannot create outgoing channel of type  
> [sofia] cause: [DESTINATION_OUT_OF_ORDER]
> 2008-12-30 14:45:25 [INFO] mod_dptools.c:1891  
> audio_bridge_function() Originate Failed.  Cause:  
> DESTINATION_OUT_OF_ORDER
>
> any suggestions?



From e.schmidbauer at gmail.com  Tue Dec 30 12:16:58 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 15:16:58 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
Message-ID: <2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>

tried  'pa call sip:886 at taz.bkw.org:5080'

2008-12-30 15:15:42 [NOTICE] switch_channel.c:565 switch_channel_set_name()
New Channel portaudio/sip:886 at taz.bkw.org:5080[a220653c-d6ae-11dd-af32-1ff260d7b236]
2008-12-30 15:15:42 [NOTICE] mod_portaudio.c:1586 place_call() Channel
[portaudio/sip:886 at taz.bkw.org:5080] has been answered
2008-12-30 15:15:42 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
FreeSwitch->sip:886 at taz.bkw.org:5080 in context default
2008-12-30 15:15:42 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid
Profile
2008-12-30 15:15:42 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close
Channel N/A [CS_NEW]
2008-12-30 15:15:42 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate()
Cannot create outgoing channel of type [sofia] cause:
[DESTINATION_OUT_OF_ORDER]
2008-12-30 15:15:42 [INFO] mod_dptools.c:1891 audio_bridge_function()
Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER
2008-12-30 15:15:42 [NOTICE] mod_dptools.c:1918 audio_bridge_function()
Hangup portaudio/sip:886 at taz.bkw.org:5080 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]

still got errors....


On Tue, Dec 30, 2008 at 3:05 PM, Brian West <brian at freeswitch.org> wrote:

> Yes actually call sip:886 at taz.bkw.org:5080   I set it up specifically
> for you to test :P
>
> /b
>
> On Dec 30, 2008, at 1:50 PM, e schmidbauer wrote:
>
> > i did as your said and got some errors in the console....
> > after i run pa call sip:886 at ww2.bwrl.org:5080 i get.....
> > 2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()
> > Processing relaxxplayer->sip:886 at ww2.bwrl.org:5080 in context default
> > 2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel()
> > Invalid Profile
> > 2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540
> > sofia_outgoing_channel() Close Channel N/A [CS_NEW]
> > 2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116
> > switch_ivr_originate() Cannot create outgoing channel of type
> > [sofia] cause: [DESTINATION_OUT_OF_ORDER]
> > 2008-12-30 14:45:25 [INFO] mod_dptools.c:1891
> > audio_bridge_function() Originate Failed.  Cause:
> > DESTINATION_OUT_OF_ORDER
> >
> > any suggestions?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Dec 30 12:32:11 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 14:32:11 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
Message-ID: <DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>

You don't have the default config ... did you modify the dialplan to  
have the propler sip_uri extension.. you only had to change one line.


On Dec 30, 2008, at 2:16 PM, e schmidbauer wrote:

> 2008-12-30 15:15:42 [ERR] mod_sofia.c:2423 sofia_outgoing_channel()  
> Invalid Profile
> 2008-12-30 15:15:42 [NOTICE] mod_sofia.c:2540  
> sofia_outgoing_channel() Close Channel N/A [CS_NEW]



From jaugenstine at gmail.com  Tue Dec 30 12:36:40 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Tue, 30 Dec 2008 12:36:40 -0800
Subject: [Freeswitch-users]  LUA execute response question
Message-ID: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>

I am developing a Freeswitch/LUA script.  From the script, I have a
requirement to retrieve information via HTTP from an app server.  I have
utilized the HTTP application from the Freeswitch CLI.  It works great.  My
question is how can I call the "http get http://www......." from within the
LUA script and retrieve the HTTP response?  Is this feasible?  Or is there a
way to make an HTTP request directly from LUA?

Jonathan
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From e.schmidbauer at gmail.com  Tue Dec 30 12:43:22 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 15:43:22 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
	<DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
Message-ID: <2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>

success! i was able to connect to your box using the CELT codec. thanks for
your help.
could you tell me how your end is configured so i can try this on my box?

On Tue, Dec 30, 2008 at 3:32 PM, Brian West <brian at freeswitch.org> wrote:

> You don't have the default config ... did you modify the dialplan to
> have the propler sip_uri extension.. you only had to change one line.
>
>
> On Dec 30, 2008, at 2:16 PM, e schmidbauer wrote:
>
> > 2008-12-30 15:15:42 [ERR] mod_sofia.c:2423 sofia_outgoing_channel()
> > Invalid Profile
> > 2008-12-30 15:15:42 [NOTICE] mod_sofia.c:2540
> > sofia_outgoing_channel() Close Channel N/A [CS_NEW]
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Dec 30 12:47:28 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 14:47:28 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
	<DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
	<2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>
Message-ID: <041099B7-5F37-4A65-9C4E-E20776094410@freeswitch.org>

Its setup to answer and playback... with CELT at 48000h allowed on the  
sofia profile.. playing a stream from my DirecTV (XM Radio 20on20 @ 48k)

btw how did it sound?

/b

On Dec 30, 2008, at 2:43 PM, e schmidbauer wrote:

> success! i was able to connect to your box using the CELT codec.  
> thanks for your help.
> could you tell me how your end is configured so i can try this on my  
> box?



From e.schmidbauer at gmail.com  Tue Dec 30 12:54:26 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 15:54:26 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <041099B7-5F37-4A65-9C4E-E20776094410@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
	<DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
	<2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>
	<041099B7-5F37-4A65-9C4E-E20776094410@freeswitch.org>
Message-ID: <2cef777b0812301254v77d3c2e5iaf9e8af400fdd148@mail.gmail.com>

sound quality is amazing like im listening to music on my own computer. can
you show me the dialplan config for the extension?

On Tue, Dec 30, 2008 at 3:47 PM, Brian West <brian at freeswitch.org> wrote:

> Its setup to answer and playback... with CELT at 48000h allowed on the
> sofia profile.. playing a stream from my DirecTV (XM Radio 20on20 @ 48k)
>
> btw how did it sound?
>
> /b
>
> On Dec 30, 2008, at 2:43 PM, e schmidbauer wrote:
>
> > success! i was able to connect to your box using the CELT codec.
> > thanks for your help.
> > could you tell me how your end is configured so i can try this on my
> > box?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Tue Dec 30 12:57:19 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 15:57:19 -0500
Subject: [Freeswitch-users] LUA execute response question
In-Reply-To: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
References: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
Message-ID: <EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>

you can execute the freeswitch api command like any other api command  
or use any loadable lua module available to do this.


Mike

On Dec 30, 2008, at 3:36 PM, jonathan augenstine wrote:

> I am developing a Freeswitch/LUA script.  From the script, I have a  
> requirement to retrieve information via HTTP from an app server.  I  
> have utilized the HTTP application from the Freeswitch CLI.  It  
> works great.  My question is how can I call the "http get http:// 
> www......." from within the LUA script and retrieve the HTTP  
> response?  Is this feasible?  Or is there a way to make an HTTP  
> request directly from LUA?

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From brian at freeswitch.org  Tue Dec 30 12:59:43 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 14:59:43 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301254v77d3c2e5iaf9e8af400fdd148@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
	<DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
	<2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>
	<041099B7-5F37-4A65-9C4E-E20776094410@freeswitch.org>
	<2cef777b0812301254v77d3c2e5iaf9e8af400fdd148@mail.gmail.com>
Message-ID: <E391462D-CA01-488C-A923-15B02674F86A@freeswitch.org>

     <extension name="xmradio">
       <condition field="destination_number" expression="^886$">
         <action application="answer"/>
         <action application="playback" data="local_stream://xmradio"/>
       </condition>
     </extension>


and local stream is just pointed at a shout cast server on my Mac that  
is plugged into my DirecTV receiver

/b


On Dec 30, 2008, at 2:54 PM, e schmidbauer wrote:

> sound quality is amazing like im listening to music on my own  
> computer. can you show me the dialplan config for the extension?



From jaugenstine at gmail.com  Tue Dec 30 13:16:36 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Tue, 30 Dec 2008 13:16:36 -0800
Subject: [Freeswitch-users] LUA execute response question
In-Reply-To: <EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>
References: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
	<EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>
Message-ID: <207e7a5e0812301316k2c7ba5dft5ba7e1c51795301a@mail.gmail.com>

Mike,

I executed the following:

>From CLI:
http get http://www.google.com {}

This comes back with the HTTP response and prints it to the console.  I
tried to execute the following:

response = session:execute("http", "get http://www.google.com {}");

What I am unable to figure out how to do is retrieve HTTP response.  That is
my dilemma.

Jonathan

On Tue, Dec 30, 2008 at 12:57 PM, Michael Jerris <mike at jerris.com> wrote:

> you can execute the freeswitch api command like any other api command or
> use any loadable lua module available to do this.
>
> Mike
>
> On Dec 30, 2008, at 3:36 PM, jonathan augenstine wrote:
>
> I am developing a Freeswitch/LUA script.  From the script, I have a
> requirement to retrieve information via HTTP from an app server.  I have
> utilized the HTTP application from the Freeswitch CLI.  It works great.  My
> question is how can I call the "http get http://www......." from within
> the LUA script and retrieve the HTTP response?  Is this feasible?  Or is
> there a way to make an HTTP request directly from LUA?
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Dec 30 13:40:45 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 16:40:45 -0500
Subject: [Freeswitch-users] LUA execute response question
In-Reply-To: <207e7a5e0812301316k2c7ba5dft5ba7e1c51795301a@mail.gmail.com>
References: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
	<EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>
	<207e7a5e0812301316k2c7ba5dft5ba7e1c51795301a@mail.gmail.com>
Message-ID: <413D08F3-9803-450A-B9B8-EE8A7F965CBF@jerris.com>

session:execute runs a freeswitch app, you want to be executing the  
fsapi command using API:execute
Mike


On Dec 30, 2008, at 4:16 PM, jonathan augenstine wrote:

> Mike,
>
> I executed the following:
>
> From CLI:
> http get http://www.google.com {}
>
> This comes back with the HTTP response and prints it to the  
> console.  I tried to execute the following:
>
> response = session:execute("http", "get http://www.google.com {}");
>
> What I am unable to figure out how to do is retrieve HTTP response.   
> That is my dilemma.
>
> Jonathan
>
> On Tue, Dec 30, 2008 at 12:57 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> you can execute the freeswitch api command like any other api  
> command or use any loadable lua module available to do this.
>
>
> Mike
>
> On Dec 30, 2008, at 3:36 PM, jonathan augenstine wrote:
>
>> I am developing a Freeswitch/LUA script.  From the script, I have a  
>> requirement to retrieve information via HTTP from an app server.  I  
>> have utilized the HTTP application from the Freeswitch CLI.  It  
>> works great.  My question is how can I call the "http get http:// 
>> www......." from within the LUA script and retrieve the HTTP  
>> response?  Is this feasible?  Or is there a way to make an HTTP  
>> request directly from LUA?

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From dyfet at gnutelephony.org  Tue Dec 30 13:48:05 2008
From: dyfet at gnutelephony.org (David Sugar)
Date: Tue, 30 Dec 2008 16:48:05 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar - a
 cute hack
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670233BC664C@mse17be1.mse17.exchange.ms>
References: <495A124C.3040006@laposte.net>
	<495A1D2D.3070507@gnutelephony.org>	<495A22E0.3040904@networkoblivion.com>
	<6E8D2069C08AA84A83D336E996AE4C670233BC664C@mse17be1.mse17.exchange.ms>
Message-ID: <495A9715.4020201@gnutelephony.org>

I actually have found an alternate approach that we optionally use in
sipwitch.  Basically, sipwitch can be set to recognize a "trusted"
subnet, and automatically accepts a refresh from any actively registered
ua on the trusted subnet(s) without requesting an authentication
challenge, so long as the ua refreshes from the same sip port and ip
address it originally registered and authenticated from.  It will also
do the same for invites and other otherwise "authentication challenge"
sip requests that can originate from ua's on the trusted subnet(s).

Using this option of course kills any ability to proxy register multiple
ua's through another sip server, although this can be solved by
recognizing certain id's as explicitly not trustable.  However, for most
common configurations and use cases, it works very well and does
effectively halve sip network traffic :).

Michael Giagnocavo wrote:
>>>  This is
>>> because the ua sends it's registration refresh unauthenticated.  The
>>> registrar will then push back an authentication challenge request so the
>>> ua can prove its identity, at which point the ua then repeats the same
>>> transaction, but with authentication credentials attached.
>> Why does it do that?  Every time I do a debug, I see the first request
>> denied as unauthorized and then it always comes right back and gets
> 
> Welcome to HTTP Digest authentication. The request has to get challenged to get a new nonce from the server (so as to mitigate replay attacks).
> 
> You could TLS and auth off of the client cert, except few devices support that, and you'd have the "overhead" of TCP (which is like bad or something).
> 
> -Michael
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dyfet at gnutelephony.org  Tue Dec 30 13:50:30 2008
From: dyfet at gnutelephony.org (David Sugar)
Date: Tue, 30 Dec 2008 16:50:30 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <7b197bef0812300527x5073b212j38b1a60f475440f6@mail.gmail.com>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
	<7b197bef0812300527x5073b212j38b1a60f475440f6@mail.gmail.com>
Message-ID: <495A97A6.1060504@gnutelephony.org>

Well, there are worse virus's one could be infected with, I suppose ;).

Actually recently I had been surviving focusing on secure VoIP and
wireless...

Giovanni Maruzzelli wrote:
> Hi David,
> 
> very happy to read you on the FS list!
> 
> We met in 2001 at OSCon San Diego, where you "infected" me with the
> telephony virus :-).
> 
> You did great work with the Bayonne project, really breaking new ground.
> 
> Thank you,
> 
> happy hacking,
> 
> happy new year!!!!
> 
> 
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> =========================================
> Company : Celliax
> Website: www.celliax.org
> Address : via Pierlombardo 9, 20135 Milano
> Country/Territory : Italy
> Business Email: gmaruzz at celliax dot org
> Cell : 39-347-2665618
> Fax : 39-02-87390039
> 
> 
> 
> 
> On Tue, Dec 30, 2008 at 2:07 PM, David Sugar <dyfet at gnutelephony.org> wrote:
>> You actually have potentially ~1320 effective "SIP transactions" per
>> second to support 40000 registered ua's with a 60s refresh.  This is
>> because the ua sends it's registration refresh unauthenticated.  The
>> registrar will then push back an authentication challenge request so the
>> ua can prove its identity, at which point the ua then repeats the same
>> transaction, but with authentication credentials attached.
>>
>> rod wrote:
>>> Hi all,
>>>
>>> I know that freeswitch has not been designed as a pure sip
>>> proxy/registrar, but I'm wondering how many subscribers could be handled
>>> by FS.
>>>
>>> I setup the following test environment:
>>>     - Kamailio 1.4.2 as the registrar
>>>     - all invite requests are flowing through FS, even for a call
>>> between 2 registered subscribers. Many reasons for this: the calls CDR
>>> are centralized in the same format, I can easily add a billing ID to a
>>> call, proceed to recording, set the caller as anonymous if requested...
>>>     - FS is used also as a SBC
>>>
>>> There is still a lot of work to do, mainly on the call forwarding
>>> feature and this is why I'm wondering (simply out of curiosity) what
>>> could have been achieved using only FS (easier to setup when only one
>>> equipment is involved :) ).
>>>
>>> I'd like to register 40 000 subscribers (if each user registers every
>>> 60s, you have approx 670 registration per second, this setup is working
>>> on Kamailio).
>>>
>>> I did the following to increase FS performance regarding registration:
>>>     - put the directory containing users in a RAMDISK
>>>     - put the db directory in a RAMDISK
>>>
>>> with this I was able to reach 190 registration per second (50 without
>>> the ramdisk) but for one SIP account, not too useful :p (for your
>>> information I see a huge improvement when switching from 1.0.1 phoenix:
>>> 150cps to FS svn 105xx: 190)
>>> When trying with 25000 SIP accounts, I got no more than 30cps.
>>>
>>> Then I tried to use the odbc mysql for registration, using this I was
>>> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these
>>> tests, the presence support has been disabled.
>>>
>>> As the IO performance seems to be a bottleneck, I'd like to know if
>>> there is a way to store the registration in memory only without database
>>> persistency.
>>>
>>> This thread is there only to share tips, not to complain about FS poor
>>> performance as a SIP registrar when compared to Kamailio. If I compare
>>> FS to a commercial SBC I'm using in production, I have to say that FS is
>>> really a great piece of software (lacks only statistics module, snmp,
>>> and heartbeat redundancy for failover).
>>>
>>> regards,
>>> rod
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jaugenstine at gmail.com  Tue Dec 30 15:31:07 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Tue, 30 Dec 2008 15:31:07 -0800
Subject: [Freeswitch-users] LUA execute response question
In-Reply-To: <413D08F3-9803-450A-B9B8-EE8A7F965CBF@jerris.com>
References: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
	<EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>
	<207e7a5e0812301316k2c7ba5dft5ba7e1c51795301a@mail.gmail.com>
	<413D08F3-9803-450A-B9B8-EE8A7F965CBF@jerris.com>
Message-ID: <207e7a5e0812301531x7cdff10bmf59c0c2e79381814@mail.gmail.com>

Mike,

Thank you.  I was confused and it is working now.

Jonathan


On Tue, Dec 30, 2008 at 1:40 PM, Michael Jerris <mike at jerris.com> wrote:

> session:execute runs a freeswitch app, you want to be executing the fsapi
> command using API:executeMike
>
>
> On Dec 30, 2008, at 4:16 PM, jonathan augenstine wrote:
>
> Mike,
>
> I executed the following:
>
> From CLI:
> http get http://www.google.com {}
>
> This comes back with the HTTP response and prints it to the console.  I
> tried to execute the following:
>
> response = session:execute("http", "get http://www.google.com {}");
>
> What I am unable to figure out how to do is retrieve HTTP response.  That
> is my dilemma.
>
> Jonathan
>
> On Tue, Dec 30, 2008 at 12:57 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> you can execute the freeswitch api command like any other api command or
>> use any loadable lua module available to do this.
>>
>> Mike
>>
>> On Dec 30, 2008, at 3:36 PM, jonathan augenstine wrote:
>>
>> I am developing a Freeswitch/LUA script.  From the script, I have a
>> requirement to retrieve information via HTTP from an app server.  I have
>> utilized the HTTP application from the Freeswitch CLI.  It works great.  My
>> question is how can I call the "http get http://www......." from within
>> the LUA script and retrieve the HTTP response?  Is this feasible?  Or is
>> there a way to make an HTTP request directly from LUA?
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fvillarroel at yahoo.com  Tue Dec 30 18:10:59 2008
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Tue, 30 Dec 2008 18:10:59 -0800 (PST)
Subject: [Freeswitch-users] New
Message-ID: <196421.77116.qm@web34307.mail.mud.yahoo.com>

Dear All,

I am new on this list, i am chilean.

I come from Asterisk and my company is dedicated to wholesale, i am provider for chile mobile.

We will need implement FreeSwictch before Asterisk for Retail and Wholesale

How i will can starting, please recomended any how to or pdf for beginning.

Is possible administrator FS from any GUI, but no from Centos Image FS,I like Debian. 

Other questions, FreeSwitch is more complex that Asterisk for begining?

Thanks everyone and excuse me English.

Fernando



      


From jason at jasonjgw.net  Tue Dec 30 18:43:48 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 31 Dec 2008 13:43:48 +1100
Subject: [Freeswitch-users] New
In-Reply-To: <196421.77116.qm@web34307.mail.mud.yahoo.com>
References: <196421.77116.qm@web34307.mail.mud.yahoo.com>
Message-ID: <20081231024348.GA10230@jdc.jasonjgw.net>

FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:
 
> How i will can starting, please recomended any how to or pdf for beginning.

http://wiki.freeswitch.org/
> 
> 
> Other questions, FreeSwitch is more complex that Asterisk for begining?

No, just different from asterisk.



From intralanman at freeswitch.org  Wed Dec 31 06:19:52 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 31 Dec 2008 14:19:52 +0000
Subject: [Freeswitch-users] New
In-Reply-To: <196421.77116.qm@web34307.mail.mud.yahoo.com>
References: <196421.77116.qm@web34307.mail.mud.yahoo.com>
Message-ID: <495B7F88.3010301@freeswitch.org>

FERNANDO VILLARROEL wrote:
> Dear All,
>
> I am new on this list, i am chilean.
>
> I come from Asterisk and my company is dedicated to wholesale, i am provider for chile mobile.
>
> We will need implement FreeSwictch before Asterisk for Retail and Wholesale
>
> How i will can starting, please recomended any how to or pdf for beginning.
>   
A couple of good starting points are:
http://wiki.freeswitch.org/wiki/Installation_Guide
http://wiki.freeswitch.org/wiki/Getting_Started_Guide

Since those are written in only english at the present time, you might 
want to join the irc channel, it's #freeswitch on irc.freenode.net ... 
there are lots of languages spoken in the channel, so someone could 
probably help you more than those guides.
> Is possible administrator FS from any GUI, but no from Centos Image FS,I like Debian. 
>   
Right now there are no GUIs completed, but several being worked on.
> Other questions, FreeSwitch is more complex that Asterisk for begining?
>   
Depending on your background, you might find it harder, or you might 
find it easier... FreeSWITCH configs are XML based where Asterisk 
configs are INI based. That's only the start of the differences.. I'd 
say give them both a try and see which shoe fits best :-)
> Thanks everyone and excuse me English.
>   
Your english isn't that bad, thanks for posting.

-Ray


From Prometheus001 at gmx.net  Wed Dec 31 06:28:32 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 31 Dec 2008 15:28:32 +0100
Subject: [Freeswitch-users] uuid_playback
Message-ID: <495B8190.2080207@gmx.net>

As I see on the Wiki page uuid_playback seems to be implemented, however
it doesn't work on the console or via event_socket.
Also in the code I could not find it (svn 10438).

So for now I use uuid_brodcast to play announcements to one or both parties.

Question: What is the status of uuid_playback?

Best regards
Peter



From gmaruzz at celliax.org  Wed Dec 31 06:58:08 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 31 Dec 2008 15:58:08 +0100
Subject: [Freeswitch-users] mod_skypiax inching forward
Message-ID: <7b197bef0812310658u2cea5f0p7e81ea099af9de83@mail.gmail.com>

Hi FreeSWITCHers!

mod_skypiax, the Skype compatible endpoint, is slowly inching toward release :-)

When the demo is online (will go on and off for development), you can
test it (so helping finding bugs) by calling with Skype the Skype
Names:

skypiax20, skypiax19, skypiax18, ...., skypiax1

Happy New Year !!!

Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039


From javieraristizabal at gmail.com  Wed Dec 31 08:57:33 2008
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Wed, 31 Dec 2008 11:57:33 -0500
Subject: [Freeswitch-users] New
In-Reply-To: <196421.77116.qm@web34307.mail.mud.yahoo.com>
References: <196421.77116.qm@web34307.mail.mud.yahoo.com>
Message-ID: <b6f11f300812310857w38c01d9fy5345ea32d647ac23@mail.gmail.com>

Hola Fernando, llevo algun tiempo trabajando con Freeswitch, si algo te
puedo ayudar me lo puedes comentar. A traves de este medio o a traves del
IRC mi nick es "javar".
Felices Fiestas.


Javier.
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From kristian.kielhofner at gmail.com  Wed Dec 31 10:44:56 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 13:44:56 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was Sonus
	and RFC 2833 related
Message-ID: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>

Hey everyone,

  A few days ago I was reading about a bug with Sonus gear and
FreeSwitch.  I remember it because Tony implemented a workaround the
buggy Sonus gear based on the SDP, all while adding a snarky (yet
hilarious) log message.

  You guessed it, I'm having 2833 timestamp problems with Sonus...

  I can't find this issue for the life of me in Jira.  Does anyone
know where it is?  It was fixed in rev 10744 with the commit message
"sonus, sonus, sonus sonus is a four letter word".

Thanks!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From brian at freeswitch.org  Wed Dec 31 11:23:23 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 31 Dec 2008 13:23:23 -0600
Subject: [Freeswitch-users] uuid_playback
In-Reply-To: <495B8190.2080207@gmx.net>
References: <495B8190.2080207@gmx.net>
Message-ID: <F952536A-A68B-4F3C-A670-0F005201FAAC@freeswitch.org>

Wiki link please.

/b

On Dec 31, 2008, at 8:28 AM, Peter P GMX wrote:

> As I see on the Wiki page uuid_playback seems to be implemented,  
> however
> it doesn't work on the console or via event_socket.
> Also in the code I could not find it (svn 10438).
>
> So for now I use uuid_brodcast to play announcements to one or both  
> parties.
>
> Question: What is the status of uuid_playback?
>
> Best regards
> Peter



From brian at freeswitch.org  Wed Dec 31 11:24:46 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 31 Dec 2008 13:24:46 -0600
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
Message-ID: <52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>

I would recommend getting the latest SVN since we had to break out the  
cisco and sonus rtp issues... Try this http://wiki.freeswitch.org/wiki/RTP_Issues

/b

On Dec 31, 2008, at 12:44 PM, Kristian Kielhofner wrote:

> Hey everyone,
>
>  A few days ago I was reading about a bug with Sonus gear and
> FreeSwitch.  I remember it because Tony implemented a workaround the
> buggy Sonus gear based on the SDP, all while adding a snarky (yet
> hilarious) log message.
>
>  You guessed it, I'm having 2833 timestamp problems with Sonus...
>
>  I can't find this issue for the life of me in Jira.  Does anyone
> know where it is?  It was fixed in rev 10744 with the commit message
> "sonus, sonus, sonus sonus is a four letter word".
>
> Thanks!



From kristian.kielhofner at gmail.com  Wed Dec 31 12:14:04 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 15:14:04 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
Message-ID: <2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>

On Wed, Dec 31, 2008 at 2:24 PM, Brian West <brian at freeswitch.org> wrote:
> I would recommend getting the latest SVN since we had to break out the
> cisco and sonus rtp issues... Try this http://wiki.freeswitch.org/wiki/RTP_Issues
>
> /b
>

Brian,

  Thanks, I have and I was well aware of these (I've been lurking).

  However, my issue is with another platform and Sonus.  I want to
demonstrate that this is a *known* issue with Sonus gear.  Plus, I
found it:

http://jira.freeswitch.org/browse/FSCORE-251

Thanks again!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From mike at jerris.com  Wed Dec 31 12:33:37 2008
From: mike at jerris.com (Michael Jerris)
Date: Wed, 31 Dec 2008 15:33:37 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
Message-ID: <43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>

If your looking for specifics of where they are being stupid and/or  
violating rfc's pop on by and let us know, we can probably detail some  
stuff that is not in the bugs as well.

Mike

On Dec 31, 2008, at 3:14 PM, Kristian Kielhofner wrote:

> On Wed, Dec 31, 2008 at 2:24 PM, Brian West <brian at freeswitch.org>  
> wrote:
>> I would recommend getting the latest SVN since we had to break out  
>> the
>> cisco and sonus rtp issues... Try this http://wiki.freeswitch.org/wiki/RTP_Issues
>>
>> /b
>>
>
> Brian,
>
>  Thanks, I have and I was well aware of these (I've been lurking).
>
>  However, my issue is with another platform and Sonus.  I want to
> demonstrate that this is a *known* issue with Sonus gear.  Plus, I
> found it:
>
> http://jira.freeswitch.org/browse/FSCORE-251
>
> Thanks again!
>
> -- 
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kristian.kielhofner at gmail.com  Wed Dec 31 13:01:10 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 16:01:10 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
	<43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>
Message-ID: <2d9149cd0812311301h2b632e45g5534aba7011c809f@mail.gmail.com>

On 12/31/08, Michael Jerris <mike at jerris.com> wrote:
> If your looking for specifics of where they are being stupid and/or
>  violating rfc's pop on by and let us know, we can probably detail some
>  stuff that is not in the bugs as well.
>
>  Mike

Mike,

  Thanks but I (think) I basically figured it out...

  I am using G.729 and RFC 2833 DTMF.

  If you send a 2833 event and voice packet with the same timestamp
Sonus will drop the event (maybe both) EVEN IF they have different
sequence numbers.  Nevermind that this behavior (same timestamps) is
quite possible and even desired in some cases and violates both RFC
1889/3550 and RFC 2833/4733.

  I didn't dig into the code but it looks like the Freeswitch
workaround just tweaks the timestamps to get around this if a Sonus
originator is detected (by parsing the SDP).  Nice.

  Did I get it?

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From anthony.minessale at gmail.com  Wed Dec 31 13:02:05 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Dec 2008 15:02:05 -0600
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
Message-ID: <191c3a030812311302w2545d099la837885e37ee412d@mail.gmail.com>

In case you want to know the nitty gritty.

excerpt from switch_types.h

RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833 = (1 << 1)
    /*
      Sonus wrongly expects that, when sending a multi-packet 2833 DTMF
event, The sender
      should increment the RTP timestamp in each packet when, in reality,
the sender should
      send the same exact timestamp and increment the duration field in the
2833 payload.
      This allows a reconstruction of the duration if any of the packets are
lost.

      final_duration - initial_timestamp = total_samples

      However, if the duration value exceeds the space allocated (16 bits),
The sender should increment
      the timestamp one unit and reset the duration to 0.

      Always sending a duration of 0 with a new timestamp should be
tolerated but is rarely intentional
      and is mistakenly done by many devices.
      The issue is that the Sonus expects everyone to do it this way instead
of tolerating either way.
      Sonus will actually ignore every packet with the same timestamp before
concluding if it's DTMF.

      This flag will cause each packet to have a new timestamp.
    */



On Wed, Dec 31, 2008 at 2:14 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On Wed, Dec 31, 2008 at 2:24 PM, Brian West <brian at freeswitch.org> wrote:
> > I would recommend getting the latest SVN since we had to break out the
> > cisco and sonus rtp issues... Try this
> http://wiki.freeswitch.org/wiki/RTP_Issues
> >
> > /b
> >
>
> Brian,
>
>  Thanks, I have and I was well aware of these (I've been lurking).
>
>  However, my issue is with another platform and Sonus.  I want to
> demonstrate that this is a *known* issue with Sonus gear.  Plus, I
> found it:
>
> http://jira.freeswitch.org/browse/FSCORE-251
>
> Thanks again!
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristian.kielhofner at gmail.com  Wed Dec 31 13:06:49 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 16:06:49 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <191c3a030812311302w2545d099la837885e37ee412d@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
	<191c3a030812311302w2545d099la837885e37ee412d@mail.gmail.com>
Message-ID: <2d9149cd0812311306i13777b93y503852bd2386865d@mail.gmail.com>

On 12/31/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> In case you want to know the nitty gritty.
>
> excerpt from switch_types.h
>
> RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833 = (1 << 1)
>     /*
>       Sonus wrongly expects that, when sending a multi-packet 2833 DTMF
> event, The sender
>        should increment the RTP timestamp in each packet when, in reality,
> the sender should
>       send the same exact timestamp and increment the duration field in the
> 2833 payload.
>       This allows a reconstruction of the duration if any of the packets are
> lost.
>
>       final_duration - initial_timestamp = total_samples
>
>       However, if the duration value exceeds the space allocated (16 bits),
> The sender should increment
>       the timestamp one unit and reset the duration to 0.
>
>       Always sending a duration of 0 with a new timestamp should be
> tolerated but is rarely intentional
>       and is mistakenly done by many devices.
>       The issue is that the Sonus expects everyone to do it this way instead
> of tolerating either way.
>        Sonus will actually ignore every packet with the same timestamp
> before concluding if it's DTMF.
>
>       This flag will cause each packet to have a new timestamp.
>     */
>

Thanks Anthony!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From anthony.minessale at gmail.com  Wed Dec 31 13:15:28 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Dec 2008 15:15:28 -0600
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <2d9149cd0812311301h2b632e45g5534aba7011c809f@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
	<43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>
	<2d9149cd0812311301h2b632e45g5534aba7011c809f@mail.gmail.com>
Message-ID: <191c3a030812311315v4fa295d3rda7b1db1ce68b85d@mail.gmail.com>

in case you want to know the other big annoying one:

If you suddenly change timestamp base mid call, sonus will lose 2 sec of
audio trying to adjust.

Say you have an ivr that asks you to dial an ext then places the call.
While the ivr is the far end and interacting with the sonus there is a
series of timestamps generated by FS.
Then when the call is placed we start passing through the timestamps from
the new farther far end so the jitter can
be preserved.  (if we rewrite the timestamps to our original ones, any
jitter would be locked in place).

With sonus, you pretty much have to set FS to rewrite the timestamps or live
with the audio drop =(




On Wed, Dec 31, 2008 at 3:01 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On 12/31/08, Michael Jerris <mike at jerris.com> wrote:
> > If your looking for specifics of where they are being stupid and/or
> >  violating rfc's pop on by and let us know, we can probably detail some
> >  stuff that is not in the bugs as well.
> >
> >  Mike
>
> Mike,
>
>  Thanks but I (think) I basically figured it out...
>
>  I am using G.729 and RFC 2833 DTMF.
>
>  If you send a 2833 event and voice packet with the same timestamp
> Sonus will drop the event (maybe both) EVEN IF they have different
> sequence numbers.  Nevermind that this behavior (same timestamps) is
> quite possible and even desired in some cases and violates both RFC
> 1889/3550 and RFC 2833/4733.
>
>  I didn't dig into the code but it looks like the Freeswitch
> workaround just tweaks the timestamps to get around this if a Sonus
> originator is detected (by parsing the SDP).  Nice.
>
>  Did I get it?
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristian.kielhofner at gmail.com  Wed Dec 31 13:35:48 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 16:35:48 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <191c3a030812311315v4fa295d3rda7b1db1ce68b85d@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
	<43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>
	<2d9149cd0812311301h2b632e45g5534aba7011c809f@mail.gmail.com>
	<191c3a030812311315v4fa295d3rda7b1db1ce68b85d@mail.gmail.com>
Message-ID: <2d9149cd0812311335k2ad2595ew2840f6076d37cb3c@mail.gmail.com>

On 12/31/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> in case you want to know the other big annoying one:
>
> If you suddenly change timestamp base mid call, sonus will lose 2 sec of
> audio trying to adjust.
>
> Say you have an ivr that asks you to dial an ext then places the call.
>  While the ivr is the far end and interacting with the sonus there is a
> series of timestamps generated by FS.
> Then when the call is placed we start passing through the timestamps from
> the new farther far end so the jitter can
>  be preserved.  (if we rewrite the timestamps to our original ones, any
> jitter would be locked in place).
>
> With sonus, you pretty much have to set FS to rewrite the timestamps or live
> with the audio drop =(
>

Anthony,

  Thanks for the pointer.  It's nice to also know that I won't be able
to implement *proper* jitter buffering in my network as long as Sonus
is involved somewhere in the call path.

  Sheesh.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From anthony.minessale at gmail.com  Wed Dec 31 14:47:11 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Dec 2008 16:47:11 -0600
Subject: [Freeswitch-users] uuid_playback
In-Reply-To: <495B8190.2080207@gmx.net>
References: <495B8190.2080207@gmx.net>
Message-ID: <191c3a030812311447t59f1e093n143b50c7fa77ca3f@mail.gmail.com>

there is no such thing as uuid_playback
broadcast is the correct and only way besides maybe uuid_displace



On Wed, Dec 31, 2008 at 8:28 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> As I see on the Wiki page uuid_playback seems to be implemented, however
> it doesn't work on the console or via event_socket.
> Also in the code I could not find it (svn 10438).
>
> So for now I use uuid_brodcast to play announcements to one or both
> parties.
>
> Question: What is the status of uuid_playback?
>
> Best regards
> Peter
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From fvillarroel at yahoo.com  Wed Dec 31 16:37:55 2008
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Wed, 31 Dec 2008 16:37:55 -0800 (PST)
Subject: [Freeswitch-users] New
In-Reply-To: <b6f11f300812310857w38c01d9fy5345ea32d647ac23@mail.gmail.com>
Message-ID: <628881.28122.qm@web34304.mail.mud.yahoo.com>

Hello

Thanks all for your comments and specially to Javier and Raymond.

Ya me tendras haciendote consultas Javier.

Regards

Fernando


--- On Wed, 12/31/08, Javier Aristiz?bal <javieraristizabal at gmail.com> wrote:

> From: Javier Aristiz?bal <javieraristizabal at gmail.com>
> Subject: Re: [Freeswitch-users] New
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, December 31, 2008, 12:57 PM
> Hola Fernando, llevo algun tiempo trabajando con Freeswitch,
> si algo te
> puedo ayudar me lo puedes comentar. A traves de este medio
> o a traves del
> IRC mi nick es "javar".
> Felices Fiestas.
> 
> 
> Javier.
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


      


From jason at jasonjgw.net  Wed Dec 31 23:20:45 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Jan 2009 18:20:45 +1100
Subject: [Freeswitch-users] fs_cli help command
Message-ID: <20090101072045.GA12582@jdc.jasonjgw.net>

I just noticed, as confirmed by reading the code, that now when the user types
"help" at the fs_cli prompt, the fs_cli help text is printed; but if what one
really wants is to execute the API help command, there doesn't seem to be any
way to do it.

process_command() gets the help command first, and there's no way to have it
passed to FreeSWITCH as an API command.

Here are a few options for solving this (I'm sure there are others):

1. Make the API command processing into a separate function, and have the
"help" command call it to execute "api help" after printing its own text. I
suppose one could also use a goto for this, but that could be accused of
inelegance.

2. Rename the fs_cli help command to "/help" - probably confusing to new
users!

3. Add a /help command that runs "api help".

I'm leaning toward option 3, but opinions may differ.



From tleyden at branchcut.com  Mon Dec  1 00:31:28 2008
From: tleyden at branchcut.com (Traun Leyden)
Date: Mon, 1 Dec 2008 13:01:28 +0430
Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 29, Issue 189
In-Reply-To: <mailman.1445.1228105114.18013.freeswitch-users@lists.freeswitch.org>
References: <mailman.1445.1228105114.18013.freeswitch-users@lists.freeswitch.org>
Message-ID: <f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>

>
> Message: 9
> Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST)
> From: Marc Orenberg <marc at kasteris.com>
> Subject: [Freeswitch-users] Problem importing modules in mod_python
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <195670.44941.qm at web50805.mail.re2.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> In the latter versions of mod_python, I'm unable to import standard python
> modules such as time and MySQLdb.?
> For example, the following script works fine in version 1.0.1:
>
> ??? ??? import time
> ??? ??? import os
> ??? ??? from freeswitch import *
> ??? ??? def handler(session, args):
> ??? ??? ??? session.answer()
> ??? ??? ??? session.execute("sleep", "2000")
> ??? ??? ???
> session.streamFile("/usr/local/freeswitch/prompts/01Welcome.wav")
> ??? ??? ??? return(session)
>
> But in freeswitch-1.0.latest.tar.gz, and svn versions 10556-10558, I get
> the following error:
>
> ??? ??? 2008-11-30 21:13:09 [ERR] mod_python.c:129 eval_some_python() Error
> reloading module
> ??? ??? Traceback (most recent call last):
> ??????? File "/usr/lib/python2.4/site-packages/scripts/test.py", line 1, in
> ?
> ??? ??? import time
> ??? ??? ImportError: /usr/lib/python2.4/lib-dynload/timemodule.so:
> undefined symbol: PyExc_ValueError
>
> Thanks for your help!
>

I have run into the same problem and put some documentation on the wiki:

http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2F...2Fdatetime.so:_undefined_symbol:_PyExc_IOError

I think something changed in freeswitch in the way it is loading modules, or
at least the way it is loading mod_python.  This behavior appeared all of
the sudden in recent freeswitch versions.

HTH
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From yudha2008 at gmail.com  Mon Dec  1 02:54:25 2008
From: yudha2008 at gmail.com (Baskar)
Date: Mon, 1 Dec 2008 16:24:25 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
Message-ID: <d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>

Hi,
 *
   **It is possible to dial outbound through console dialing. Yes means me
How ?**

   Without using the softphone how can i dial outbound from freeswitch
console itself. *
*
   I want to Know without using any softphone for calling.

   It is possible in asterisk. we can dial from console itself. *

*   So i want to know it is possible in freeswitch.*

Warm Regards,
N.Baskar
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From gmaruzz at celliax.org  Mon Dec  1 03:15:34 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 1 Dec 2008 12:15:34 +0100
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
Message-ID: <7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>

Hello Baskar,

in FS it is possible to call from console using the endpoint mod_portaudio.

Please have a look at
http://wiki.freeswitch.org/wiki/Freeswitch_softphone , it is *NOT REAL
SOFTPHONE* it is FS used *LIKE* a softphone. Exactly as in Asterisk
with chan_alsa or chan_oss.

Sincerely,

Giovanni Maruzzelli

=========================================
Contact person : Mr Giovanni Maruzzelli
Company : celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Phone : 39-347-2665618
Fax : 39-02-87390039



On Mon, Dec 1, 2008 at 11:54 AM, Baskar <yudha2008 at gmail.com> wrote:
> Hi,
>
>    It is possible to dial outbound through console dialing. Yes means me How
> ?
>
>    Without using the softphone how can i dial outbound from freeswitch
> console itself.
>
>    I want to Know without using any softphone for calling.
>
>    It is possible in asterisk. we can dial from console itself.
>
>    So i want to know it is possible in freeswitch.
>
> Warm Regards,
> N.Baskar
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From Prometheus001 at gmx.net  Mon Dec  1 03:39:06 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 01 Dec 2008 12:39:06 +0100
Subject: [Freeswitch-users] Inbound 1-way audio issue using GSM codec
In-Reply-To: <000301c951ba$9705dac0$c5119040$@com>
References: <49305CBF.8060801@ieee.org> <000301c951ba$9705dac0$c5119040$@com>
Message-ID: <4933CCDA.40905@gmx.net>

Hello Maxim,

can you reach another internal device except the GSM one in order to see
whether it's GSM codec specific?

However I can see that you're using local IPs (10.x.x.x) so I expect
that they are natted. This often causes one way audio when the external
rtp-ip is not set. Please try to set a
    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
entry to internal.xml and external.xml in your SIP profiles and see if
it works. Use stun at least for the internal profile (FQDN and external
IP most probably will not work)

Best regards
Peter

Maxim Karp schrieb:
> Hello,
>
> I am using a GSM based endpoint connected to freeswitch that makes calls to
> the PSTN via a SIP gateway (SBC).  The SBC uses PCMU between itself and
> freeswitch.
>
> When I make an outgoing call from a GSM based device via freewsitch to the
> PSTN via the SBC, everything works fine and audio works in both directions
> for both end points.  I looked at the console logs and they do indicate that
> I am using GSM.
>
> Console output when I dial and before answer on the GSM device:
>
> v=0
> o=- 74 0 IN IP4 10.229.0.58
> s=session
> c=IN IP4 10.229.0.58
> b=CT:17
> t=0 0
> m=audio 59806 RTP/AVP 8 0 3 97 101
> k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 RED/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=encryption:optional
>
> Console output once it rings and after I answer on the PSTN side:
>
> v=0
> o=FreeSWITCH 1227887572 1227887573 IN IP4 10.229.0.10
> s=FreeSWITCH
> c=IN IP4 10.229.0.10
> t=0 0
> a=sendrecv
> m=audio 30896 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> When I receive a call from the SIP gateway, the endpoint making the call
> (not on freeswitch) can't hear me speaking from the GSM device connected to
> freeswitch.  I can hear everything fine on the GSM device.
>
> Here is the console output for the call info coming in from the PSTN.
>
> v=0
> o=FreeSWITCH 1227902084 1227902085 IN IP4 38.113.164.132
> s=FreeSWITCH
> c=IN IP4 38.113.164.132
> t=0 0
> a=sendrecv
> m=audio 16724 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> Here is how I have vars.xml configured:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
>
> When I prioritize GSM on the outbound codec prefs I get static on the PSTN
> side. 
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/>
>
> Any ideas?
>
> Maxim.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From odermann at googlemail.com  Mon Dec  1 03:46:39 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 1 Dec 2008 12:46:39 +0100
Subject: [Freeswitch-users] Leg A terminated by Leg B on a uuid_bridge
In-Reply-To: <872970CF4A55BF42A5337D570860209F01052E34@HPEXCHVS01.exchange.airg>
References: <872970CF4A55BF42A5337D570860209F01052E34@HPEXCHVS01.exchange.airg>
Message-ID: <5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>

hi  simon,

i am not sure, if i understood your problem right, but if you do not
want leg a to hang up after leg b (the originated call) hangs up, set
"park_after_bridge=true" when you make the originate.
as far as i know, "hangup_after_bridge=false" is only for the inbound
and helps nothing with the outbound.

if you want something different, please explain me a little more.

dennis



2008/11/28 Simon Tang <simon at airg.com>:
> Hello,
>
>
>
> I'm using event socket outbound, and have an issue where, after a bridge
> ends and is terminated by Leg B, Leg A is also terminated.  Here's the call
> flow:
>
>
>
> 1.       Call comes in (Leg A), session created, play welcome message.
>
> 2.       From this session, originate and dial out using api originate
>
> 3.       After the target answers (Leg B), bridge the 2 calls using api
> uuid_bridge
>
> 4.       Leg B hangs up.
>
> 5.       Leg A will be terminated.
>
>
>
> After step 4, Leg A is terminated.  I do not want Leg A to hang up.  I've
> tried setting "hangup_after_bridge=false" prior to the call, and that
> doesn't work.
>
>
>
> Having said that, I tried a similar test which does not end Leg A's call
> after Leg B hangs up, but I can't use this solution because, functionally,
> does not accomplish what I want it to do (i.e., I want to perform some
> actions on Leg B prior to the bridge, like send some DTMF tones, playback
> some messages, etc).  I did not need to set the "hangup_after_bridge"
> variable (default should be false anyway).
>
>
>
> 1.       Call comes in (Leg A), session created, play welcome message.
>
> 2.       From this session, do a bridge by doing an execute bridge.
>
> 3.       The target answers (Leg B)
>
> 4.       Leg B hangs up.
>
> 5.       Leg A will still be active.
>
>
>
> Any ideas would be appreciated.  Thanks!
>
>
>
> Simon
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From saigop at gmail.com  Mon Dec  1 04:25:31 2008
From: saigop at gmail.com (Gopala krishnan)
Date: Mon, 1 Dec 2008 17:55:31 +0530
Subject: [Freeswitch-users] Leg A terminated by Leg B on a uuid_bridge
In-Reply-To: <5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>
References: <872970CF4A55BF42A5337D570860209F01052E34@HPEXCHVS01.exchange.airg>
	<5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>
Message-ID: <2ea4d47e0812010425i1f278768i2be711c74a2e00b8@mail.gmail.com>

Hi Simon,
  You can get the A leg uuid and B leg uuid seperately and can hangup
whichever the leg you need...:)



-- 
Thank you  with regards,
Gopal,
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From ttroy50 at gmail.com  Mon Dec  1 05:09:57 2008
From: ttroy50 at gmail.com (matrim)
Date: Mon, 1 Dec 2008 05:09:57 -0800 (PST)
Subject: [Freeswitch-users]  TLS receiving calls
Message-ID: <20771637.post@talk.nabble.com>


Hi,

I'm having problems using TLS to receive calls.

I'm using a Nokia N95 to test TLS against freeswitch. I can register my
client against freeswitch and make outbound calls to the test numbers (e.g.
9999).

I can also make calls to other users registered over UDP.

However if I try to make a call to a user registered over TLS the leg of the
call to that user always goes via UDP.

e.g.

1000 registered via TLS
1001 registered via TLS
1002 registered via UDP
1003 registered via UDP

1000 -> 1002 works ok
1003 -> 1002 works ok

1001 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000
tries to setup via UDP
1002 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000
tries to setup via UDP

===

>From looking at some of the documentation it seems to me that the issue may
be with the "tls-bind-params" being "transport=tls".

The phone I'm using doesn't add the "transport=tls" parameter, and only uses
"sips:" to specify that the connection is via TLS.

I tried setting "tls-bind-params" to a blank string but it didn't change
anything. Is there any way to receive calls over TLS if you don't specify
"transport=tls" in your contact string during registration?

According to RFC3261 the use of the "transport=tls" parameter isn't
recommended anymore and is now deprecated.


-- 
View this message in context: http://www.nabble.com/TLS-receiving-calls-tp20771637p20771637.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From fs_ask_sy at citromail.hu  Mon Dec  1 05:35:39 2008
From: fs_ask_sy at citromail.hu (x y)
Date: Mon, 01 Dec 2008 14:35:39 +0100
Subject: [Freeswitch-users] Multi FS behind same NAT,
	PRACK goes to wrong port
In-Reply-To: <5FD8C155-AFD9-451E-B58D-31CC47CB2EA6@freeswitch.org>
Message-ID: <20081201133539.27255.qmail@server15.citromail.hu>

            Hy!

You were right about the contact in 183, its port 5060 in there.
I've tried turning of 100rel, it seemed to work with calls, but caused some problems with others things, so I would really appreciate if there is another option.
Btw, I have mentioned that, that I had gateway problems too. Setting up ext-ip as stun.freeswitch.org has seemed to work, but after 5 days, the gateway has went down again with the same 503 error. Is there any common in the two issues?

Thx for your advices.

Cheers,
Viktor





################################################################
U xxx.xxx.xxx.xxx:56965 -&gt; yyy.yyy.yyy.yyy:5060
INVITE sip:252252%233619995384 at box.net:5060 SIP/2.0.
Via: SIP/2.0/UDP  xxx .xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" &lt;sip:xxx.xxx.xxx.xxx&gt;;tag=7185D258-BB0.
To: &lt;sip:252252%233619995384 at box.net&gt;.
Date: Thu, 27 Nov 2008 16:28:41 GMT.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313 at xxx.xxx.xxx.xxx.
Supported: timer,100rel.
Min-SE:  1800.
Cisco-Guid: 1234720477-3151434205-2374282417-4237312787.
User: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO.
CSeq: 101 INVITE.
Max-Forwards: 10.
Remote-Party-ID: &lt;sip:xxx.xxx.xxx.xxx&gt;;party=calling;screen=yes;privacy=full.
Timestamp: 1227803321.
Contact: &lt;sip:xxx.xxx.xxx.xxx:5060&gt;.
Expires: 180.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 264.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 5202 8450 IN IP4 xxx.xxx.xxx.xxx.
s=SIP Call.
c=IN IP4 xxx.xxx.xxx.xxx.
t=0 0.
m=audio 16732 RTP/AVP 3 8 101.
c=IN IP4 xxx.xxx.xxx.xxx.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.

#
U yyy.yyy.yyy.yyy:5060 -&gt; xxx.xxx.xxx.xxx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" &lt;sip:xxx.xxx.xxx.xxx&gt;;tag=7185D258-BB0.
To: &lt;sip:252252%233619995384 at box.net&gt;.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313 at xxx.xxx.xxx.xxx.
CSeq: 101 INVITE.
Timestamp: 1227803321 0.000388.
User-Agent: agent
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -&gt; zzz.zzz.zzz.zzz:1352
INVITE sip:3619995384 at zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F.
Max-Forwards: 8.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
Contact: &lt;sip:usr at yyy.yyy.yyy.yyy:5060;transport=udp&gt;.
User-Agent: agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary.
Min-SE: 120.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 398.
Remote-Party-ID: "00000000" &lt;sip:00000000 at dom&gt;;screen=yes;privacy=full.
.
v=0.
o=FreeSWITCH 6476130113585053783 8141266268953030291 IN IP4 yyy.yyy.yyy.yyy.
s=FreeSWITCH.
c=IN IP4 yyy.yyy.yyy.yyy.
t=0 0.
a=sendrecv.
m=audio 17068 RTP/AVP 3 98 8 9 0 18 101 13.
a=rtpmap:3 GSM/8000.
a=rtpmap:98 SPEEX/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U zzz.zzz.zzz.zzz:1352 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
User-Agent: agent2
Content-Length: 0.
.

###
U zzz.zzz.zzz.zzz:1352 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
Contact: &lt;sip:mod_sofia at zzz.zzz.zzz.zzz:5060;transport=udp&gt;.
RSeq: 2093511444.
User-Agent: agent2
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Require: 100rel.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 264.
.
v=0.
o=FreeSWITCH 6247558966294607749 119302723364474833 IN IP4 zzz.zzz.zzz.zzz.
s=FreeSWITCH.
c=IN IP4 zzz.zzz.zzz.zzz.
t=0 0.
m=audio 24756 RTP/AVP 3 101 13.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U yyy.yyy.yyy.yyy:5060 -&gt; zzz.zzz.zzz.zzz:5060
PRACK sip:mod_sofia at zzz.zzz.zzz.zzz:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK5DH0ryBH70FSB.
Max-Forwards: 70.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783325 PRACK.
Contact: &lt;sip:usr at yyy.yyy.yyy.yyy:5060;transport=udp&gt;.
RAck: 2093511444 107783324 INVITE .
User-Agent: agent
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Content-Length: 0.
.

#
U zzz.zzz.zzz.zzz:5060 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 481 No such response.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5DH0ryBH70FSB.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783325 PRACK.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -&gt; zzz.zzz.zzz.zzz:1352
CANCEL sip:3619995384 at zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F .
Max-Forwards: 8.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 CANCEL.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -&gt; xxx.xxx.xxx.xxx:5060
SIP/2.0 481 Call/Transaction Does Not Exist.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" &lt;sip:xxx.xxx .xxx.xxx&gt;;tag=7185D258-BB0.
To: &lt;sip:252252%233619995384 at box.net&gt;;tag=cXQ5m2QSmU2QB.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313 at xxx .xxx.xxx.xxx.
CSeq: 101 INVITE.
User-Agent: agent
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: 100rel, timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary.
Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE".
Content-Length: 0.
.

#
U xxx.xxx.xxx.xxx:56965 -&gt; yyy.yyy.yyy.yyy:5060
ACK sip:252252%233619995384 at box.net:5060 SIP/2.0.
Via: SIP/2.0/UDP  xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:eTel".
From: "box" &lt;sip:xxx.xxx.xxx.xxx&gt;;tag=7185D258-BB0.
To: &lt;sip:252252%233619995384 at box.net&gt;;tag=cXQ5m2QSmU2QB.
Date: Thu, 27 Nov 2008 16:28:41 GMT.
Call-ID: 499A2BEE-BBD711DD-8D87ACB1-FC904313 at xxx.xxx.xxx.xxx.
Max-Forwards: 10.
Content-Length: 0.
CSeq: 101 ACK.
.

#
U zzz.zzz.zzz.zzz:1352 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 CANCEL.
Content-Length: 0.
.

#
U zzz.zzz.zzz.zzz:1352 -&gt; yyy.yyy.yyy.yyy:5060
SIP/2.0 487 Request Terminated.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK44Q7p3tDarS6F.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 INVITE.
User-Agent: agent2
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Length: 0.
.

#
U yyy.yyy.yyy.yyy:5060 -&gt; zzz.zzz.zzz.zzz:1352
ACK sip:3619995384 at zzz.zzz.zzz.zzz:1352 SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy;rport;branch=z9hG4bK44Q7p3tDarS6F.
Max-Forwards: 8.
From: "00000000" &lt;sip:usr at dom;transport=udp&gt;;tag=D6gypX8vH4raQ.
To: &lt;sip:3619995384 at zzz.zzz.zzz.zzz:1352&gt;;tag=yaQQt10jBBg5H.
Call-ID: 4ab334d0-3743-122c-1c91-00e081349397.
CSeq: 107783324 ACK.
Content-Length: 0.
.

#########################################################





Hirdet?s (x)      


RENDELJ MOST! - H?ztart?si g?peket AKCI?S ?ron!


T?bb mint 300 VIDE? term?kbemutat? seg?t v?lasztani, ak?r 5 ?V kiterjesztett garanci?val rendelhetsz ITT!  

S?t?-f?z?lap szettek, mos?g?pek, mosogat?g?pek, t?zhelyek - ORSZ?GOS sz?ll?t?ssal a
MARKABOLT.hu-t?l.
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From regs at kinetix.gr  Mon Dec  1 06:01:40 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Mon, 01 Dec 2008 16:01:40 +0200
Subject: [Freeswitch-users] Set variable for the outgoing leg
Message-ID: <4933EE44.60900@kinetix.gr>

All the variables that I set show up only in the a-leg CDR.
How can I set a variable that can be used during the b-leg CDR generation?



From Prometheus001 at gmx.net  Mon Dec  1 07:47:00 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 01 Dec 2008 16:47:00 +0100
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <20771637.post@talk.nabble.com>
References: <20771637.post@talk.nabble.com>
Message-ID: <493406F4.204@gmx.net>

Did you add
<action application="export" data="sip_secure_media=true"/>
into youy dialplan before bridging that call. How is your internal.conf,
is TLS enabled there?

Best regards
Peter

matrim schrieb:
> Hi,
>
> I'm having problems using TLS to receive calls.
>
> I'm using a Nokia N95 to test TLS against freeswitch. I can register my
> client against freeswitch and make outbound calls to the test numbers (e.g.
> 9999).
>
> I can also make calls to other users registered over UDP.
>
> However if I try to make a call to a user registered over TLS the leg of the
> call to that user always goes via UDP.
>
> e.g.
>
> 1000 registered via TLS
> 1001 registered via TLS
> 1002 registered via UDP
> 1003 registered via UDP
>
> 1000 -> 1002 works ok
> 1003 -> 1002 works ok
>
> 1001 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000
> tries to setup via UDP
> 1002 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000
> tries to setup via UDP
>
> ===
>
>   
>> >From looking at some of the documentation it seems to me that the issue may
>>     
> be with the "tls-bind-params" being "transport=tls".
>
> The phone I'm using doesn't add the "transport=tls" parameter, and only uses
> "sips:" to specify that the connection is via TLS.
>
> I tried setting "tls-bind-params" to a blank string but it didn't change
> anything. Is there any way to receive calls over TLS if you don't specify
> "transport=tls" in your contact string during registration?
>
> According to RFC3261 the use of the "transport=tls" parameter isn't
> recommended anymore and is now deprecated.
>
>
>   



From brian at freeswitch.org  Mon Dec  1 07:59:42 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2008 09:59:42 -0600
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <493406F4.204@gmx.net>
References: <20771637.post@talk.nabble.com> <493406F4.204@gmx.net>
Message-ID: <D6CA81AB-3B46-4A33-8076-18838D3392AA@freeswitch.org>

sip_secure_media only activates SRTP.

/b

On Dec 1, 2008, at 9:47 AM, Peter P GMX wrote:

> Did you add
> <action application="export" data="sip_secure_media=true"/>
> into youy dialplan before bridging that call. How is your  
> internal.conf,
> is TLS enabled there?
>
> Best regards
> Peter




From brian at freeswitch.org  Mon Dec  1 08:00:25 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2008 10:00:25 -0600
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <20771637.post@talk.nabble.com>
References: <20771637.post@talk.nabble.com>
Message-ID: <EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>

Please tell that to everyone out there in the REAL world.  It was my  
understanding that sips: was the one that went away in favor of  
transport= which is what everyone uses.

/b

On Dec 1, 2008, at 7:09 AM, matrim wrote:

> According to RFC3261 the use of the "transport=tls" parameter isn't
> recommended anymore and is now deprecated.




From mkarp at securesilence.com  Mon Dec  1 08:24:04 2008
From: mkarp at securesilence.com (Maxim Karp)
Date: Mon, 1 Dec 2008 08:24:04 -0800
Subject: [Freeswitch-users] Inbound 1-way audio issue using GSM codec
In-Reply-To: <4933CCDA.40905@gmx.net>
References: <49305CBF.8060801@ieee.org> <000301c951ba$9705dac0$c5119040$@com>
	<4933CCDA.40905@gmx.net>
Message-ID: <003e01c953d1$3b6f25e0$b24d71a0$@com>

Hi Peter,

Thanks for your response.

When I use PCMU two-way audio works fine.

When I make outgoing calls from a Freeswitch extension (using GSM) and then
out to a gateway using PCMU everything works fine.

When I receive calls from the same gateway, the end point behind the gateway
can't hear me.

The GSM-PSMU (and viceversa) transcoding for outgoing from an endpoint
associated with a  Freeswitch  extension to the external gateway is perfect
but incoming there seems to be an issue.

Maxim.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P
GMX
Sent: December-01-08 3:39 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inbound 1-way audio issue using GSM codec

Hello Maxim,

can you reach another internal device except the GSM one in order to see
whether it's GSM codec specific?

However I can see that you're using local IPs (10.x.x.x) so I expect
that they are natted. This often causes one way audio when the external
rtp-ip is not set. Please try to set a
    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
entry to internal.xml and external.xml in your SIP profiles and see if
it works. Use stun at least for the internal profile (FQDN and external
IP most probably will not work)

Best regards
Peter

Maxim Karp schrieb:
> Hello,
>
> I am using a GSM based endpoint connected to freeswitch that makes calls
to
> the PSTN via a SIP gateway (SBC).  The SBC uses PCMU between itself and
> freeswitch.
>
> When I make an outgoing call from a GSM based device via freewsitch to the
> PSTN via the SBC, everything works fine and audio works in both directions
> for both end points.  I looked at the console logs and they do indicate
that
> I am using GSM.
>
> Console output when I dial and before answer on the GSM device:
>
> v=0
> o=- 74 0 IN IP4 10.229.0.58
> s=session
> c=IN IP4 10.229.0.58
> b=CT:17
> t=0 0
> m=audio 59806 RTP/AVP 8 0 3 97 101
> k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 RED/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=encryption:optional
>
> Console output once it rings and after I answer on the PSTN side:
>
> v=0
> o=FreeSWITCH 1227887572 1227887573 IN IP4 10.229.0.10
> s=FreeSWITCH
> c=IN IP4 10.229.0.10
> t=0 0
> a=sendrecv
> m=audio 30896 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> When I receive a call from the SIP gateway, the endpoint making the call
> (not on freeswitch) can't hear me speaking from the GSM device connected
to
> freeswitch.  I can hear everything fine on the GSM device.
>
> Here is the console output for the call info coming in from the PSTN.
>
> v=0
> o=FreeSWITCH 1227902084 1227902085 IN IP4 38.113.164.132
> s=FreeSWITCH
> c=IN IP4 38.113.164.132
> t=0 0
> a=sendrecv
> m=audio 16724 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> Here is how I have vars.xml configured:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
>
> When I prioritize GSM on the outbound codec prefs I get static on the PSTN
> side. 
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/>
>
> Any ideas?
>
> Maxim.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Mon Dec  1 08:26:27 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 10:26:27 -0600
Subject: [Freeswitch-users] Multi FS behind same NAT,
	PRACK goes to wrong port
In-Reply-To: <20081128142036.4190.qmail@server15.citromail.hu>
References: <20081128142036.4190.qmail@server15.citromail.hu>
Message-ID: <191c3a030812010826y20d60707o2b2a30973fbd7e11@mail.gmail.com>

if you enable nat mode on the registrations it will lock the ip:port
make an acl that matches the ip of the client  and add the param
apply-nat-acl with the name of the acl you created to your sofia profile
then all calls from that ip will be known to be nat and the port locking
code will activate.



On Fri, Nov 28, 2008 at 8:20 AM, x y <fs_ask_sy at citromail.hu> wrote:

> Hy!
>
> There are two different FS behind the same NAT, and there were Reigstration
> Failures about one or to times a day. The gateway status turned down, then I
> got 503 error codes. Then I set up the ext-ip to STUN, as the wiki requests
> it.
> Now I facing the next problem:
> Start the call, all goes right, INVITE goes to port 1352, then after 183
> Session progress from port 1352, the PRACK package goes to 5060 instead of
> 1352, wich messes up the call procedure. Is there anyway to force PRACK to
> the port to the INVITE has been sent before?
>
> Cheers,
> Viktor
>
>
>
> *Hirdet?s (x) *
> V?ltson most olcs?bb k?telez?re a biztos?t?s-hu-val. www.biztositas.hu<http://ad.adverticum.net/b/cl,1,2201,295040,360471/click.prm>- a k?telez? biztos?t?sok kiindul?pontja!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Mon Dec  1 08:27:38 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 10:27:38 -0600
Subject: [Freeswitch-users] Set variable for the outgoing leg
In-Reply-To: <4933EE44.60900@kinetix.gr>
References: <4933EE44.60900@kinetix.gr>
Message-ID: <191c3a030812010827r64f6c66er8d3a5e49868430d6@mail.gmail.com>

if you use the export instead of set app then they will get set on both
legs.
otherwise vars you only want set on b leg you can add to the dial string
{foo=bar,test=true}sofia/default/user at dest.com


On Mon, Dec 1, 2008 at 8:01 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:

> All the variables that I set show up only in the a-leg CDR.
> How can I set a variable that can be used during the b-leg CDR generation?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Mon Dec  1 08:29:11 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 10:29:11 -0600
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <586215.20730.qm@web30701.mail.mud.yahoo.com>
References: <586215.20730.qm@web30701.mail.mud.yahoo.com>
Message-ID: <191c3a030812010829v5860336ah35960774ee5d5af1@mail.gmail.com>

404 not found means the extension you are dialing is not found, not the
sound file the extension is playing.
press f8 and try again and the debug log will help you figure it out.


On Mon, Dec 1, 2008 at 12:09 AM, Faisal Maqsoodi
<faisalmaqsoodi at yahoo.com>wrote:

> I tried to play a sound file using the dialplan given on the link
> http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav
>
> <action application="playback" data="/path/to/your.wav"/>
> In place of /path/to/your.wave I used
> "/en/us/callie/misc/8000/call_secured.wav"
> "/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
> "/sounds/en/us/callie/misc/8000/call_secured.wav"
> But none of these is useful bcoz when i call on 2009, which is
> to b dialed to play the sound, same msg is
> displayed "404 NOT FOUND"
> Plz help me out.    Faisal
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From ttroy50 at gmail.com  Mon Dec  1 08:42:30 2008
From: ttroy50 at gmail.com (Thomas Troy)
Date: Mon, 1 Dec 2008 16:42:30 +0000
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <493406F4.204@gmx.net>
References: <20771637.post@talk.nabble.com> <493406F4.204@gmx.net>
Message-ID: <adbe60280812010842t21283046ndd0d0734844eb738@mail.gmail.com>

I don't have that set however I'm not trying to use SRTP yet. At the moment
I'm just trying to use Secure SIP.

That section of my dial plan is

      <condition field="${sip_has_crypto}"
expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
break="never">
        <action application="set" data="sip_secure_media=true"/>
        <!-- Offer SRTP on outbound legs if we have it on inbound. -->
        <!-- <action application="export" data="sip_secure_media=true"/> -->
      </condition>

The TLS part of my internal.xml is now

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="true"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value=""/>
    <!-- Port to listen on for TLS requests. (5061 will be used if
unspecified) -->
    <param name="tls-sip-port" value="$${internal_tls_port}"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed
for TLS server) -->
    <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not
work with TLSv1 -->
    <param name="tls-version" value="$${sip_tls_version}"/>


I also tried with

    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>

On Mon, Dec 1, 2008 at 3:47 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Did you add
> <action application="export" data="sip_secure_media=true"/>
> into youy dialplan before bridging that call. How is your internal.conf,
> is TLS enabled there?
>
> Best regards
> Peter
>
> matrim schrieb:
> > Hi,
> >
> > I'm having problems using TLS to receive calls.
> >
> > I'm using a Nokia N95 to test TLS against freeswitch. I can register my
> > client against freeswitch and make outbound calls to the test numbers
> (e.g.
> > 9999).
> >
> > I can also make calls to other users registered over UDP.
> >
> > However if I try to make a call to a user registered over TLS the leg of
> the
> > call to that user always goes via UDP.
> >
> > e.g.
> >
> > 1000 registered via TLS
> > 1001 registered via TLS
> > 1002 registered via UDP
> > 1003 registered via UDP
> >
> > 1000 -> 1002 works ok
> > 1003 -> 1002 works ok
> >
> > 1001 -> 1000 Doesn't work. The leg of the call between freeswitch and
> 1000
> > tries to setup via UDP
> > 1002 -> 1000 Doesn't work. The leg of the call between freeswitch and
> 1000
> > tries to setup via UDP
> >
> > ===
> >
> >
> >> >From looking at some of the documentation it seems to me that the issue
> may
> >>
> > be with the "tls-bind-params" being "transport=tls".
> >
> > The phone I'm using doesn't add the "transport=tls" parameter, and only
> uses
> > "sips:" to specify that the connection is via TLS.
> >
> > I tried setting "tls-bind-params" to a blank string but it didn't change
> > anything. Is there any way to receive calls over TLS if you don't specify
> > "transport=tls" in your contact string during registration?
> >
> > According to RFC3261 the use of the "transport=tls" parameter isn't
> > recommended anymore and is now deprecated.
> >
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ttroy50 at gmail.com  Mon Dec  1 08:49:43 2008
From: ttroy50 at gmail.com (Thomas Troy)
Date: Mon, 1 Dec 2008 16:49:43 +0000
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
Message-ID: <adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>

I'm not sure about current implementations that servers are using. I'm used
to using sip over UDP and TCP but this is my first time testing SIP over
TLS. So I'm just going by what's in the specification and what's implemented
on the devices I'm trying to test against, which are Nokia S60 devices (e.g.
Nokia N95, E66).

Out of interest do you have any links to anywhere this is discussed in terms
of general sip implementations?

On Mon, Dec 1, 2008 at 4:00 PM, Brian West <brian at freeswitch.org> wrote:

> Please tell that to everyone out there in the REAL world.  It was my
> understanding that sips: was the one that went away in favor of
> transport= which is what everyone uses.
>
> /b
>
> On Dec 1, 2008, at 7:09 AM, matrim wrote:
>
> > According to RFC3261 the use of the "transport=tls" parameter isn't
> > recommended anymore and is now deprecated.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Dec  1 09:02:08 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 11:02:08 -0600
Subject: [Freeswitch-users] Inbound 1-way audio issue using GSM codec
In-Reply-To: <000301c951ba$9705dac0$c5119040$@com>
References: <49305CBF.8060801@ieee.org> <000301c951ba$9705dac0$c5119040$@com>
Message-ID: <191c3a030812010902y688f6f08x5adfdd34349d4fde@mail.gmail.com>

probably pstn side has acknowledged our gsm then sent ulaw anyway and we
think its gsm.
most likely there are multiple codecs in the accept packet from the gateway
and they expect us to figure out what codec to use based on the first packet
we get from them rather than just accepting one codec in the sdp like 90% of
devices so we have a proper chance to setup optimal packetization.  This is
one of those lame parts of the RFC that describe complete unscalable
stupidity that some stuff likes to tout for who knows why.

one thing you can try is to set the variable aboslute_codec_string in the
dial to force
only gsm to be advertised at all making it impossible for the remote end to
respond with multiple codecs.

<action application="bridge"
data="{absolute_codec_string=GSM}sofia/<profile>/<uri>"/>




On Fri, Nov 28, 2008 at 6:36 PM, Maxim Karp <mkarp at securesilence.com> wrote:

> Hello,
>
> I am using a GSM based endpoint connected to freeswitch that makes calls to
> the PSTN via a SIP gateway (SBC).  The SBC uses PCMU between itself and
> freeswitch.
>
> When I make an outgoing call from a GSM based device via freewsitch to the
> PSTN via the SBC, everything works fine and audio works in both directions
> for both end points.  I looked at the console logs and they do indicate
> that
> I am using GSM.
>
> Console output when I dial and before answer on the GSM device:
>
> v=0
> o=- 74 0 IN IP4 10.229.0.58
> s=session
> c=IN IP4 10.229.0.58
> b=CT:17
> t=0 0
> m=audio 59806 RTP/AVP 8 0 3 97 101
> k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 RED/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=encryption:optional
>
> Console output once it rings and after I answer on the PSTN side:
>
> v=0
> o=FreeSWITCH 1227887572 1227887573 IN IP4 10.229.0.10
> s=FreeSWITCH
> c=IN IP4 10.229.0.10
> t=0 0
> a=sendrecv
> m=audio 30896 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> When I receive a call from the SIP gateway, the endpoint making the call
> (not on freeswitch) can't hear me speaking from the GSM device connected to
> freeswitch.  I can hear everything fine on the GSM device.
>
> Here is the console output for the call info coming in from the PSTN.
>
> v=0
> o=FreeSWITCH 1227902084 1227902085 IN IP4 38.113.164.132
> s=FreeSWITCH
> c=IN IP4 38.113.164.132
> t=0 0
> a=sendrecv
> m=audio 16724 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> Here is how I have vars.xml configured:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
>
> When I prioritize GSM on the outbound codec prefs I get static on the PSTN
> side.
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/>
>
> Any ideas?
>
> Maxim.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec  1 09:06:56 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 11:06:56 -0600
Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 29, Issue 189
In-Reply-To: <f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>
References: <mailman.1445.1228105114.18013.freeswitch-users@lists.freeswitch.org>
	<f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>
Message-ID: <191c3a030812010906m8ed33b2i6a9d65dc8b1c962f@mail.gmail.com>

we no longer use global name space in our shared objects which seems to have
a side effect on modules who in turn try to load it's own shared objects
because they too inherit the non-global namespace param.

you can either add an attribute to the modues.conf to ask it to load with
global name space

<load module="mod_foo" global="true"/>

or you can edit the code to request global loading every time.

by adding the param SMODF_GLOBAL_SYMBOLS to the SWITCH_MODULE_DEFINITION
macro
(See mod_cepstral at the top)


On Mon, Dec 1, 2008 at 2:31 AM, Traun Leyden <tleyden at branchcut.com> wrote:

>
>
>> Message: 9
>> Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST)
>> From: Marc Orenberg <marc at kasteris.com>
>> Subject: [Freeswitch-users] Problem importing modules in mod_python
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID: <195670.44941.qm at web50805.mail.re2.yahoo.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> In the latter versions of mod_python, I'm unable to import standard python
>> modules such as time and MySQLdb.?
>> For example, the following script works fine in version 1.0.1:
>>
>> ??? ??? import time
>> ??? ??? import os
>> ??? ??? from freeswitch import *
>> ??? ??? def handler(session, args):
>> ??? ??? ??? session.answer()
>> ??? ??? ??? session.execute("sleep", "2000")
>> ??? ??? ???
>> session.streamFile("/usr/local/freeswitch/prompts/01Welcome.wav")
>> ??? ??? ??? return(session)
>>
>> But in freeswitch-1.0.latest.tar.gz, and svn versions 10556-10558, I get
>> the following error:
>>
>> ??? ??? 2008-11-30 21:13:09 [ERR] mod_python.c:129 eval_some_python()
>> Error reloading module
>> ??? ??? Traceback (most recent call last):
>> ??????? File "/usr/lib/python2.4/site-packages/scripts/test.py", line 1,
>> in ?
>> ??? ??? import time
>> ??? ??? ImportError: /usr/lib/python2.4/lib-dynload/timemodule.so:
>> undefined symbol: PyExc_ValueError
>>
>> Thanks for your help!
>>
>
> I have run into the same problem and put some documentation on the wiki:
>
>
> http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2F...2Fdatetime.so:_undefined_symbol:_PyExc_IOError
>
> I think something changed in freeswitch in the way it is loading modules,
> or at least the way it is loading mod_python.  This behavior appeared all of
> the sudden in recent freeswitch versions.
>
> HTH
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
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IRC: irc.freenode.net #freeswitch

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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Dec  1 09:29:00 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 11:29:00 -0600
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
Message-ID: <191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>

yes,

mod_shout will broadcast calls as MP3 that you can listen to in
itunes/winamp live.


On Fri, Nov 28, 2008 at 8:51 AM, Dennis <odermann at googlemail.com> wrote:

> so i would have to make a call with a phone to a specific dialplan? if
> so, this would not be, what i whished (although it is nice to have the
> option).
>
> isn't there something, which can stream the voice of a given uuid? so
> i could place a link in the html admin-area to spy an uuid and to hear
> everything over the speaker? this would be really sexy ;)
>
>
> 2008/11/18 Anthony Minessale <anthony.minessale at gmail.com>:
> > you can use the eavesdrop dialplan app from a new call to spy on an in
> > progress session
> > it takes the uuid of the channel you want to listen to as the arg.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From mike at jerris.com  Mon Dec  1 09:28:34 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Dec 2008 12:28:34 -0500
Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 29, Issue 189
In-Reply-To: <f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>
References: <mailman.1445.1228105114.18013.freeswitch-users@lists.freeswitch.org>
	<f9f80d540812010031r7c1edc5ar8df0457736e467b9@mail.gmail.com>
Message-ID: <8DC3637C-9C3D-4F25-9361-06165893F116@jerris.com>

Try changing the module definition to use global symbols the same way  
we did in mod_lua, see if that resolves the issue.

Mike

On Dec 1, 2008, at 3:31 AM, Traun Leyden wrote:

>
>
> Message: 9
> Date: Sun, 30 Nov 2008 18:42:30 -0800 (PST)
> From: Marc Orenberg <marc at kasteris.com>
> Subject: [Freeswitch-users] Problem importing modules in mod_python
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <195670.44941.qm at web50805.mail.re2.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> In the latter versions of mod_python, I'm unable to import standard  
> python modules such as time and MySQLdb.?
> For example, the following script works fine in version 1.0.1:
>
> ??? ??? import time
> ??? ??? import os
> ??? ??? from freeswitch import *
> ??? ??? def handler(session, args):
> ??? ??? ??? session.answer()
> ??? ??? ??? session.execute("sleep", "2000")
> ??? ??? ??? session.streamFile("/usr/local/freeswitch/prompts/ 
> 01Welcome.wav")
> ??? ??? ??? return(session)
>
> But in freeswitch-1.0.latest.tar.gz, and svn versions 10556-10558, I  
> get the following error:
>
> ??? ??? 2008-11-30 21:13:09 [ERR] mod_python.c:129  
> eval_some_python() Error reloading module
> ??? ??? Traceback (most recent call last):
> ??????? File "/usr/lib/python2.4/site-packages/scripts/test.py",  
> line 1, in ?
> ??? ??? import time
> ??? ??? ImportError: /usr/lib/python2.4/lib-dynload/timemodule.so:  
> undefined symbol: PyExc_ValueError
>
> Thanks for your help!
>
> I have run into the same problem and put some documentation on the  
> wiki:
>
> http://wiki.freeswitch.org/wiki/Mod_python#ImportError:_.2F... 
> 2Fdatetime.so:_undefined_symbol:_PyExc_IOError
>
> I think something changed in freeswitch in the way it is loading  
> modules, or at least the way it is loading mod_python.  This  
> behavior appeared all of the sudden in recent freeswitch versions.
>
> HTH
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Mon Dec  1 09:37:45 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 11:37:45 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
Message-ID: <191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>

the guy who made mod_openmrcp has stopped development and is now making a
new library called
unimrcp it will take some time to create a new module and remove the now
unsupported openmrcp.



On Fri, Nov 28, 2008 at 12:15 PM, <mszlazak at aol.com> wrote:

> I'm getting the following errors when trying to run the example in the
> wiki: http://wiki.freeswitch.org/wiki/Mod_openmrcp
>
> 2008-11-28 09:59:54 [DEBUG] switch_core_session.c:435
> switch_core_session_receive_message() Send signal sofia/internal/
> 1000 at 10.0.0.2 [BREAK]
> 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel
> sofia/internal/1000 at 10.0.0.2 entering state [completed]
> 2008-11-28 09:59:54 [NOTICE] mod_spidermonkey.c:2034 session_answer()
> Channel [sofia/internal/1000 at 10.0.0.2] has been answered
> 2008-11-28 09:59:54 [DEBUG] mod_spidermonkey.c:1851 init_speech_engine()
> Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
> 2008-11-28 09:59:54 [DEBUG] mod_openmrcp.c:634 openmrcp_tts_open() Create
> Synthesizer Channel
> 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel
> sofia/internal/1000 at 10.0.0.2 entering state [ready]
>
> 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:643 openmrcp_tts_open() No
> response from client stack
> 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:647 openmrcp_tts_open() No
> synthesizer channel available
> 2008-11-28 09:59:59 [ERR] mod_spidermonkey.c:1859 init_speech_engine()
> Invalid TTS module!
> 2008-11-28 09:59:59 [ERR] inline:1 mod_spidermonkey()  Cannot allocate
> speech engine!
>
> 2008-11-28 09:59:59 [NOTICE] switch_core_state_machine.c:160
> switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2[CS_EXECUTE] [NORMAL_CLEARING]
> 2008-11-28 09:59:59 [DEBUG] switch_channel.c:1449
> switch_channel_perform_hangup() Send signal sofia/internal/1000 at 10.0.0.2[KILL]
> 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:727
> switch_core_session_signal_state_change() Send signal sofia/internal/
> 1000 at 10.0.0.2 [BREAK]
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:432
> switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State EXECUTE
> going to sleep
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:367
> switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) Running State
> Change CS_HANGUP
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395
> switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State HANGUP
> 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:276 sofia_on_hangup() Channel
> sofia/internal/1000 at 10.0.0.2 hanging up, cause: NORMAL_CLEARING
> 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:333 sofia_on_hangup() Sending BYE
> to sofia/internal/1000 at 10.0.0.2
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:46
> switch_core_standard_on_hangup() sofia/internal/1000 at 10.0.0.2 Standard
> HANGUP, cause: NORMAL_CLEARING
> 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395
> switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State HANGUP
> going to sleep
> 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:860
> switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2)
> Locked, Waiting on external entities
> 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:878
> switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2)
> Ended
> 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:880
> switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2[CS_HANGUP]
> 2008-11-28 10:00:26 [DEBUG] mod_openmrcp.c:167
> openmrcp_on_session_terminate() on_session_terminate called
>
> I believe I followed the instructions correctly but I can't get openmrcp to
> connect with Cepstrals TTS.
>
> ------------------------------
> Tis the season to save your money! Get the new AOL Holiday Toolbar<http://toolbar.aol.com/holiday/download.html?ncid=emlweusdown00000008>for money saving offers and gift ideas.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From simon at airg.com  Mon Dec  1 10:01:47 2008
From: simon at airg.com (Simon Tang)
Date: Mon, 1 Dec 2008 10:01:47 -0800
Subject: [Freeswitch-users] Leg A terminated by Leg B on a uuid_bridge
In-Reply-To: <5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>
References: <872970CF4A55BF42A5337D570860209F01052E34@HPEXCHVS01.exchange.airg>
	<5e414ed0812010346q6f6c59aai9d0645b3320953d0@mail.gmail.com>
Message-ID: <872970CF4A55BF42A5337D570860209F01052EA8@HPEXCHVS01.exchange.airg>

Thanks Dennis,

That did exactly what I needed.

Cheers!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Dennis
Sent: December 1, 2008 3:47 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Leg A terminated by Leg B on a
uuid_bridge

hi  simon,

i am not sure, if i understood your problem right, but if you do not
want leg a to hang up after leg b (the originated call) hangs up, set
"park_after_bridge=true" when you make the originate.
as far as i know, "hangup_after_bridge=false" is only for the inbound
and helps nothing with the outbound.

if you want something different, please explain me a little more.

dennis



2008/11/28 Simon Tang <simon at airg.com>:
> Hello,
>
>
>
> I'm using event socket outbound, and have an issue where, after a
bridge
> ends and is terminated by Leg B, Leg A is also terminated.  Here's the
call
> flow:
>
>
>
> 1.       Call comes in (Leg A), session created, play welcome message.
>
> 2.       From this session, originate and dial out using api originate
>
> 3.       After the target answers (Leg B), bridge the 2 calls using
api
> uuid_bridge
>
> 4.       Leg B hangs up.
>
> 5.       Leg A will be terminated.
>
>
>
> After step 4, Leg A is terminated.  I do not want Leg A to hang up.
I've
> tried setting "hangup_after_bridge=false" prior to the call, and that
> doesn't work.
>
>
>
> Having said that, I tried a similar test which does not end Leg A's
call
> after Leg B hangs up, but I can't use this solution because,
functionally,
> does not accomplish what I want it to do (i.e., I want to perform some
> actions on Leg B prior to the bridge, like send some DTMF tones,
playback
> some messages, etc).  I did not need to set the "hangup_after_bridge"
> variable (default should be false anyway).
>
>
>
> 1.       Call comes in (Leg A), session created, play welcome message.
>
> 2.       From this session, do a bridge by doing an execute bridge.
>
> 3.       The target answers (Leg B)
>
> 4.       Leg B hangs up.
>
> 5.       Leg A will still be active.
>
>
>
> Any ideas would be appreciated.  Thanks!
>
>
>
> Simon
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From mszlazak at aol.com  Mon Dec  1 10:19:42 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 01 Dec 2008 13:19:42 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
Message-ID: <8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>


 Hi Anthony,

Oh! OK.

So is this module "totally broken". 

I say this because I can't seem to get it to work at all with the example in that Mod_openmrcp wiki page but I thought it might because I'm not be using the right Cepstral software (freetrial download versus the paided for SDK) or that I'm not using the right port numbers or something else I didn't do. I used TcpView to look at local port associated with my Cepstral software and changed a few things but still nothing. I changed the loglevel setting to 7 in the wiki's example but I don't see the kind of output on the console that I would expect for debug mode.

Thanks. Mark.


 


 

-----Original Message-----
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 1 Dec 2008 9:37 am
Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP









the guy who made mod_openmrcp has stopped development and is now making a new library called 
unimrcp it will take some time to create a new module and remove the now unsupported openmrcp.





On Fri, Nov 28, 2008 at 12:15 PM,  <mszlazak at aol.com> wrote:


I'm getting the following errors when trying to run the example in the wiki: http://wiki.freeswitch.org/wiki/Mod_openmrcp




2008-11-28 09:59:54 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Send signal sofia/internal/1000 at 10.0.0.2 [BREAK]

2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 10.0.0.2 entering state [completed]

2008-11-28 09:59:54 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000 at 10.0.0.2] has been answered

2008-11-28 09:59:54 [DEBUG] mod_spidermonkey.c:1851 init_speech_engine() Raw Codec Activation Success L16 at 8000hz 1 channel 20ms

2008-11-28 09:59:54 [DEBUG] mod_openmrcp.c:634 openmrcp_tts_open() Create Synthesizer Channel

2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 10.0.0.2 entering state [ready]



2008-11-28 09:59:59 [ERR] mod_openmrcp.c:643 openmrcp_tts_open() No response from client stack

2008-11-28 09:59:59 [ERR] mod_openmrcp.c:647 openmrcp_tts_open() No synthesizer channel available

2008-11-28 09:59:59 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!

2008-11-28 09:59:59 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine!



2008-11-28 09:59:59 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]

2008-11-28 09:59:59 [DEBUG] switch_channel.c:1449 switch_channel_perform_hangup() Send signal sofia/internal/1000 at 10.0.0.2 [KILL]

2008-11-28 09:59:59 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal sofia/internal/1000 at 10.0.0.2 [BREAK]

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State EXECUTE going to sleep

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:367 switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) Running State Change CS_HANGUP

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State HANGUP

2008-11-28 09:59:59 [DEBUG] mod_sofia.c:276 sofia_on_hangup() Channel sofia/internal/1000 at 10.0.0.2 hanging up, cause: NORMAL_CLEARING

2008-11-28 09:59:59 [DEBUG] mod_sofia.c:333 sofia_on_hangup() Sending BYE to sofia/internal/1000 at 10.0.0.2

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1000 at 10.0.0.2 Standard HANGUP, cause: NORMAL_CLEARING

2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/1000 at 10.0.0.2) State HANGUP going to sleep

2008-11-28 09:59:59 [DEBUG] switch_core_session.c:860 switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2) Locked, Waiting on external entities

2008-11-28 09:59:59 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2) Ended

2008-11-28 09:59:59 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]

2008-11-28 10:00:26 [DEBUG] mod_openmrcp.c:167 openmrcp_on_session_terminate() on_session_terminate called



I believe I followed the instructions correctly but I can't get openmrcp to connect with Cepstrals TTS.






 Tis the season to save your money!  Get the new AOL Holiday Toolbar for money saving offers and gift ideas. 



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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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http://www.freeswitch.org







-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/


AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



 





_______________________________________________
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From sergey.kirillov at gmail.com  Mon Dec  1 07:26:58 2008
From: sergey.kirillov at gmail.com (Sergey Kirillov)
Date: Mon, 01 Dec 2008 17:26:58 +0200
Subject: [Freeswitch-users] Support for Junghanns duoBRI
Message-ID: <49340242.3040403@gmail.com>

Greetings,

Can somebody tell me, if it is possible to use duoBRI card
(http://www.junghanns.net/en/duobri_express_produkt.html) from
Junghanns.net together with Freeswitch?

I've found that this card has Zaptel drivers, and Freeswitch has
mod_openzap. On the other side, I saw somewhere in wiki that Freeswitch
does not support BRI at all at the moment.


Please confirm or allay my apprehensions.




From mike at jerris.com  Mon Dec  1 10:30:28 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Dec 2008 13:30:28 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
Message-ID: <CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>

I would not say it is totally broken, it is known to work in quite a  
few places, but we are unlikely to be doing any new fixes in it.

Mike

On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:

> Hi Anthony,
>
> Oh! OK.
>
> So is this module "totally broken".
>
> I say this because I can't seem to get it to work at all with the  
> example in that Mod_openmrcp wiki page but I thought it might  
> because I'm not be using the right Cepstral software (freetrial  
> download versus the paided for SDK) or that I'm not using the right  
> port numbers or something else I didn't do. I used TcpView to look  
> at local port associated with my Cepstral software and changed a few  
> things but still nothing. I changed the loglevel setting to 7 in the  
> wiki's example but I don't see the kind of output on the console  
> that I would expect for debug mode.
>
> Thanks. Mark.
>
>
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From mike at jerris.com  Mon Dec  1 10:31:32 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Dec 2008 13:31:32 -0500
Subject: [Freeswitch-users] Support for Junghanns duoBRI
In-Reply-To: <49340242.3040403@gmail.com>
References: <49340242.3040403@gmail.com>
Message-ID: <B22D2E3D-9C6F-43C6-A634-DC00B91BDBBB@jerris.com>

The bri support is still in development, basic calls on ptmp bri do  
appear to work, although I am not sure with what hardware.

Mike


On Dec 1, 2008, at 10:26 AM, Sergey Kirillov wrote:

> Greetings,
>
> Can somebody tell me, if it is possible to use duoBRI card
> (http://www.junghanns.net/en/duobri_express_produkt.html) from
> Junghanns.net together with Freeswitch?
>
> I've found that this card has Zaptel drivers, and Freeswitch has
> mod_openzap. On the other side, I saw somewhere in wiki that  
> Freeswitch
> does not support BRI at all at the moment.
>
>
> Please confirm or allay my apprehensions.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mszlazak at aol.com  Mon Dec  1 10:41:38 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 01 Dec 2008 13:41:38 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com><191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com><8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
Message-ID: <8CB21FB943565E5-CA4-C41@MBLK-M05.sysops.aol.com>


 

MikeJ, if openMRCP isn't totally broken then would you mind helping me get the example in Mod_openMRCP working or something like it since I don't know what the heck I'm doing wrong.

I can meet you now over at the IRC channel for Freeswitch users if you like.

Thanks.

 

-----Original Message-----
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 1 Dec 2008 10:30 am
Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP









I would not say it is totally broken, it is known to work in quite a few places, but we are unlikely to be doing any new fixes in it.



Mike




On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:



 Hi Anthony,
 
 Oh! OK.
 
 So is this module "totally broken". 
 
 I say this because I can't seem to get it to work at all with the example in that Mod_openmrcp wiki page but I thought it might because I'm not be using the right Cepstral software (freetrial download versus the paided for SDK) or that I'm not using the right port numbers or something else I didn't do. I used TcpView to look at local port associated with my Cepstral software and changed a few things but still nothing. I changed the loglevel setting to 7 in the wiki's example but I don't see the kind of output on the console that I would expect for debug mode.
 
 Thanks. Mark.
 
 
 
 
 





=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From dave at 3c.co.uk  Mon Dec  1 10:51:39 2008
From: dave at 3c.co.uk (David Knell)
Date: Mon, 01 Dec 2008 18:51:39 +0000
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
Message-ID: <4934323B.9000305@3c.co.uk>

Hi Mike,

My experience is that it's somewhat broken - it took two trivial tweaks 
to get it to work with IBM's ASR and TTS, but there's a more intractable 
problem to do with memory getting overwritten (I assume that this is 
something to do with something being freed when it shouldn't be) which 
causes a segfault on the second or third session after the module being 
loaded.

Without wishing to sound like a stuck record, one thing that you guys 
really ought to do is to decide what's supported and what isn't, and 
make this obvious - for example, move unsupported modules to a different 
place in the tree, don't have them built by default, etc.  MRCP is in 
the specsheet on the Wiki.  Otherwise folk like Mark and I spend time 
installing stuff, go round in circles a bit trying to make it work, and 
then find out (a) that it doesn't and (b) it's not going to be fixed 
because it's not supported.

Cheers --

Dave
> I would not say it is totally broken, it is known to work in quite a 
> few places, but we are unlikely to be doing any new fixes in it.
>
> Mike
>
> On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com <mailto:mszlazak at aol.com> 
> wrote:
>
>> Hi Anthony,
>>
>> Oh! OK.
>>
>> So is this module "totally broken".
>>
>> I say this because I can't seem to get it to work at all with the 
>> example in that Mod_openmrcp wiki page but I thought it might because 
>> I'm not be using the right Cepstral software (freetrial download 
>> versus the paided for SDK) or that I'm not using the right port 
>> numbers or something else I didn't do. I used TcpView to look at 
>> local port associated with my Cepstral software and changed a few 
>> things but still nothing. I changed the loglevel setting to 7 in the 
>> wiki's example but I don't see the kind of output on the console that 
>> I would expect for debug mode.
>>
>> Thanks. Mark.
>>
>>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From anthony.minessale at gmail.com  Mon Dec  1 11:17:56 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Dec 2008 13:17:56 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <4934323B.9000305@3c.co.uk>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
Message-ID: <191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>

mod_openmrcp was a contribution to the community by a 3rd party individual.

As i have clearly stated in 2 previous emails, the man has decided to
discontinue the openmrcp project.
So now we are left with the remains of the module and discontinued code.
This was not our decision it was his.

Since the author of openmrcp has stated that he has a new unimrcp we are
certainly going to
work towards getting mod_unimrcp to replace mod_openmrcp.  He had already
commented on that previous thread to state he is willing to consider making
a new module.

Some people use it without issue which may mean that the crash you reported
is windows specific and I do not have a working lab of any mrcp capbable
system to try it against in unix for that matter.  I have a list of work to
do from here to the moon and back so on an issue like this, unless someone
can hand me login credentials to some box and give me a phone number to dial
to reporduce the issue, it will be a long time until we can deal with it.
And the question arises, should we bother working on it anymore if the lib
has been abandoned and we cannot even get any support from it's author which
is where the problem most likely lies.

I try not to get too annoyed by these remarks about what we *ought to do*
because I know people lose sight of how much of the work to support the
project is done by a small group of 3 people and not the 2000 people it
appears to be from the outside looking in. (I've been answering email for 4
hours now)

My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
FS that we can maintain in tree knowing the development can be supported by
the original author.


On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:

>  Hi Mike,
>
> My experience is that it's somewhat broken - it took two trivial tweaks to
> get it to work with IBM's ASR and TTS, but there's a more intractable
> problem to do with memory getting overwritten (I assume that this is
> something to do with something being freed when it shouldn't be) which
> causes a segfault on the second or third session after the module being
> loaded.
>
> Without wishing to sound like a stuck record, one thing that you guys
> really ought to do is to decide what's supported and what isn't, and make
> this obvious - for example, move unsupported modules to a different place in
> the tree, don't have them built by default, etc.  MRCP is in the specsheet
> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
> round in circles a bit trying to make it work, and then find out (a) that it
> doesn't and (b) it's not going to be fixed because it's not supported.
>
> Cheers --
>
> Dave
>
> I would not say it is totally broken, it is known to work in quite a few
> places, but we are unlikely to be doing any new fixes in it.
>  Mike
>
>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>
>  Hi Anthony,
>
> Oh! OK.
>
> So is this module "totally broken".
>
> I say this because I can't seem to get it to work at all with the example
> in that Mod_openmrcp wiki page but I thought it might because I'm not be
> using the right Cepstral software (freetrial download versus the paided for
> SDK) or that I'm not using the right port numbers or something else I didn't
> do. I used TcpView to look at local port associated with my Cepstral
> software and changed a few things but still nothing. I changed the loglevel
> setting to 7 in the wiki's example but I don't see the kind of output on the
> console that I would expect for debug mode.
>
> Thanks. Mark.
>
>
>   ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> David Knell, Director, 3C Limited
> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gkuri at ieee.org  Mon Dec  1 11:34:03 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 01 Dec 2008 11:34:03 -0800
Subject: [Freeswitch-users] SIP INVITE timeout
In-Reply-To: <8DE97AFC-64E1-45A4-9B33-21C6300F52B4@freeswitch.org>
References: <49305CBF.8060801@ieee.org>
	<8DE97AFC-64E1-45A4-9B33-21C6300F52B4@freeswitch.org>
Message-ID: <49343C2B.3020302@ieee.org>

Brian,

Will setting progress_timeout = 8 and originate_timeout = 30 help me out
in this situation without using pre_answer?

Basically I'd like to timeout the INVITE to the phone in 8 seconds if it
doesn't respond to the INVITE (phone is not on the network) and send the
call to voicemail, but if the phone is actually ringing and no one picks
up in 30 seconds, send it to voicemail?

Thanks
Gabe

Brian West wrote:
> Try pre_answer before bridge.
> 
> /b
> 
> Sent from my iPhone
> 
> On Nov 28, 2008, at 3:03 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
> 
>> I have a phone that is registered to FS but is no longer available
>> (Internet connection down, phone turned off, etc.). The registration
>> still exists in the sip_registrations table (not expired yet), but the
>> phone is not reachable on the network.
>>
>> According to my dialplan, if the bridge to the phone fails after 20
>> seconds, the call should be forwarded to a different box for handling
>> (see dialplan below).
>>
>>
>>      <extension name="My Extension">
>>       <condition field="destination_number" expression="1213XXXXXXX">
>>             <action application="info"/>
>>        <action application="set" data="originate_timeout=20"/>
>>             <action application="set" 
>> data="hangup_after_bridge=true"/>
>>             <action application="bridge"
>> data="${sofia_contact(default/1213XXXXXXX at mydomain.net"/>
>>        <action application="bridge"
>> data="sofia/default/1213XXXXXXX at box.mydomain.net"/>
>>        </condition>
>>      </extension>
>>
>>
>> If the phone is down and not responding to the INVITEs, it appears my
>> carrier is canceling the SIP INVITE to FreeSWITCH after about 10
>> seconds. My timeout is 20 seconds. Is there anyway to deal with this
>> situation, without going back to my carrier and asking them to 
>> increase
>> their timeout on an INVITE?
>>
>> Call Progress:
>>
>> Carrier -> FS (INVITE)
>> FS -> Carrier (100 Trying)
>>
>> <10 seconds pass while FS is attempting to contact the phone>
>>
>> Carrier -> FS (CANCEL)
>> FS -> Carrier (200 OK)
>> FS -> Carrier (487 Request Terminated)
>> Carrier -> FS (ACK)
>>
>>
>> Thanks ...
>>
>> Gabe
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From msc at freeswitch.org  Mon Dec  1 11:37:27 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Dec 2008 11:37:27 -0800
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
Message-ID: <87f2f3b90812011137v6f8c9125x9b2ae7b7f5e5bc21@mail.gmail.com>

FYI,
I've updated the wiki to reflect the current status of OpenMRCP with a link
to the new UniMRCP project. Hopefully enough people who want MRCP in FS will
support UniMRCP...
-MC

On Mon, Dec 1, 2008 at 11:17 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> mod_openmrcp was a contribution to the community by a 3rd party individual.
>
> As i have clearly stated in 2 previous emails, the man has decided to
> discontinue the openmrcp project.
> So now we are left with the remains of the module and discontinued code.
> This was not our decision it was his.
>
> Since the author of openmrcp has stated that he has a new unimrcp we are
> certainly going to
> work towards getting mod_unimrcp to replace mod_openmrcp.  He had already
> commented on that previous thread to state he is willing to consider making
> a new module.
>
> Some people use it without issue which may mean that the crash you reported
> is windows specific and I do not have a working lab of any mrcp capbable
> system to try it against in unix for that matter.  I have a list of work to
> do from here to the moon and back so on an issue like this, unless someone
> can hand me login credentials to some box and give me a phone number to dial
> to reporduce the issue, it will be a long time until we can deal with it.
> And the question arises, should we bother working on it anymore if the lib
> has been abandoned and we cannot even get any support from it's author which
> is where the problem most likely lies.
>
> I try not to get too annoyed by these remarks about what we *ought to do*
> because I know people lose sight of how much of the work to support the
> project is done by a small group of 3 people and not the 2000 people it
> appears to be from the outside looking in. (I've been answering email for 4
> hours now)
>
> My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
> FS that we can maintain in tree knowing the development can be supported by
> the original author.
>
>
> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>
>>  Hi Mike,
>>
>> My experience is that it's somewhat broken - it took two trivial tweaks to
>> get it to work with IBM's ASR and TTS, but there's a more intractable
>> problem to do with memory getting overwritten (I assume that this is
>> something to do with something being freed when it shouldn't be) which
>> causes a segfault on the second or third session after the module being
>> loaded.
>>
>> Without wishing to sound like a stuck record, one thing that you guys
>> really ought to do is to decide what's supported and what isn't, and make
>> this obvious - for example, move unsupported modules to a different place in
>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>> round in circles a bit trying to make it work, and then find out (a) that it
>> doesn't and (b) it's not going to be fixed because it's not supported.
>>
>> Cheers --
>>
>> Dave
>>
>> I would not say it is totally broken, it is known to work in quite a few
>> places, but we are unlikely to be doing any new fixes in it.
>>  Mike
>>
>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>
>>  Hi Anthony,
>>
>> Oh! OK.
>>
>> So is this module "totally broken".
>>
>> I say this because I can't seem to get it to work at all with the example
>> in that Mod_openmrcp wiki page but I thought it might because I'm not be
>> using the right Cepstral software (freetrial download versus the paided for
>> SDK) or that I'm not using the right port numbers or something else I didn't
>> do. I used TcpView to look at local port associated with my Cepstral
>> software and changed a few things but still nothing. I changed the loglevel
>> setting to 7 in the wiki's example but I don't see the kind of output on the
>> console that I would expect for debug mode.
>>
>> Thanks. Mark.
>>
>>
>>   ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.kubr at gmail.com  Mon Dec  1 12:48:53 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Mon, 1 Dec 2008 21:48:53 +0100
Subject: [Freeswitch-users] Sound file as ringback
Message-ID: <698401620812011248g53ef6579q5ca03ba22ce69709@mail.gmail.com>

Yes, setting the var to the full path works. Sorry, should have taken
the "full path" in the wiki more seriously.
MP3s are played only once, 8kHz WAVs work perfectly.

Cheers,
Jan

> Can you try putting the full path to the file?  Also what does the
> console output look like?
>
> /b
>
> On Nov 30, 2008, at 12:30 PM, Jan Kubr wrote:
>
>> I have try different format of files (from 8KHz mono wavs to MP3s, all
>> of which play fine via playback) and some caused the bridge to be
>> finished immediately (with NO_USER_RESPONSE), some make it generate
>> crazy beeping, but none is played while the phone is ringing on the
>> other end.



From jan.kubr at gmail.com  Mon Dec  1 12:55:51 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Mon, 1 Dec 2008 21:55:51 +0100
Subject: [Freeswitch-users] How to specify Path for sound files
Message-ID: <698401620812011255g2e15991w4970559ac912f9b3@mail.gmail.com>

Hi Faisal,
the path is either an absolute path or a path relative to the
directory in the sound_prefix var in vars.xml.
So this

<action application="playback"
data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"/>

works fine on my box. You sure this one doesn't work for you?

Jan

> I tried to play a sound file using the dialplan given on the link
> http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav
>
> <action application="playback" data="/path/to/your.wav"/>
> In place of /path/to/your.wave I used
> "/en/us/callie/misc/8000/call_secured.wav"
> "/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
> "/sounds/en/us/callie/misc/8000/call_secured.wav"
> But none of these is useful bcoz when i call on 2009, which is
> to b dialed to play the sound, same msg is
> displayed "404 NOT FOUND"
> Plz help me out.??? Faisal



From mszlazak at aol.com  Mon Dec  1 16:44:31 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 01 Dec 2008 19:44:31 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com><191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com><8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com><CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com><4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
Message-ID: <8CB222E46498CA1-11E0-E8B@MBLK-M05.sysops.aol.com>


 


 Does "bridging" a call from FS to Voxeo's Prophecy server require openMRCP? If not then the other issue I might have is a database look up that is part of the dialogue that maybe need as the person response to prompts from the asr. It's possible to run a php script for the database stuff that Prophecy might need or could that happen via Javascript in FS? Then after the dialogue has completed I go from Prophecy back to FS.


 

-----Original Message-----
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 1 Dec 2008 11:17 am
Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP









mod_openmrcp was a contribution to the community by a 3rd party individual.

As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project.
So now we are left with the remains of the module and discontinued code.? This was not our decision it was his. 


Since the author of openmrcp has stated that he has a new unimrcp we are certainly going to 
work towards getting mod_unimrcp to replace mod_openmrcp.? He had already commented on that previous thread to state he is willing to consider making a new module.



Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter.? I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it.? And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies.


I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) 


My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author.



On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:





  



Hi Mike,



My experience is that it's somewhat broken - it took two trivial tweaks
to get it to work with IBM's ASR and TTS, but there's a more
intractable problem to do with memory getting overwritten (I assume
that this is something to do with something being freed when it
shouldn't be) which causes a segfault on the second or third session
after the module being loaded.



Without wishing to sound like a stuck record, one thing that you guys
really ought to do is to decide what's supported and what isn't, and
make this obvious - for example, move unsupported modules to a
different place in the tree, don't have them built by default, etc.?
MRCP is in the specsheet on the Wiki.? Otherwise folk like Mark and I
spend time installing stuff, go round in circles a bit trying to make
it work, and then find out (a) that it doesn't and (b) it's not going
to be fixed because it's not supported.



Cheers --



Dave






I would not say it is totally broken, it is known to work
in quite a few places, but we are unlikely to be doing any new fixes in
it.
  


  

  
Mike

  


  

  
On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:

  

  

    
 Hi Anthony,

    

Oh! OK.

    

So is this module "totally broken". 

    

I say this because I can't seem to get it to work at all with the
example in that Mod_openmrcp wiki page but I thought it might because
I'm not be using the right Cepstral software (freetrial download versus
the paided for SDK) or that I'm not using the right port numbers or
something else I didn't do. I used TcpView to look at local port
associated with my Cepstral software and changed a few things but still
nothing. I changed the loglevel setting to 7 in the wiki's example but
I don't see the kind of output on the console that I would expect for
debug mode.

    

Thanks. Mark.

    

    
 

    

    


    

  

  

  

  






_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  









-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 





_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org







-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/


AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From kkielhofner at star2star.com  Mon Dec  1 19:43:01 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Mon, 1 Dec 2008 22:43:01 -0500
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
Message-ID: <2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>

On 12/1/08, Thomas Troy <ttroy50 at gmail.com> wrote:
..snip..
>
> Out of interest do you have any links to anywhere this is discussed in terms
> of general sip implementations?
>

Uh oh, here we go again...

http://www.iana.org/assignments/sip-parameters
http://tools.ietf.org/html/rfc3969

https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-August/010047.html

Implementation wise, most devices tend to use transport=tls:

SIPFoundry - From what I've seen
Snom
SERs
Asterisk (If you are using TLS)
Cisco - I *believe* you can use either a SIPS URI or the transport=tls
parameter for various SIP targets

  As the RFC (basically) states (RFC3261, section 12.1.x),
transport=tls was deprecated in RFC 3261 because you should also be
able to do TLS over SCTP (RFC3436), which makes transport=tls a bit
ambiguous. sips:user at domain;transport=tcp or
sips:user at domain;transport=sctp is a bit more flexible.

  I don't know if I've ever seen anything default to SIPS URIs.  I
also don't think I've ever specifically tried using them.  However, my
experience with TLS is admittedly somewhat limited so this shouldn't
be taken as gospel.  As you can see from the discussions on
sip-implementors, this gets interesting when different devices are
traversing a proxy using different URI schemes...

  However, I suspect this won't become an issue until most SIP
implementations support SCTP.  That should be exciting! ;)


-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From faisalmaqsoodi at yahoo.com  Mon Dec  1 21:30:53 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Mon, 1 Dec 2008 21:30:53 -0800 (PST)
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <698401620812011255g2e15991w4970559ac912f9b3@mail.gmail.com>
Message-ID: <625821.37656.qm@web30706.mail.mud.yahoo.com>

Yes its not working on my system. When i copy this in default.xml dialplan, it works but as a seperate extension in dialplan/extensions it does'nt.
???????????????????????????????????????????????????????????????????????????????????????? Faisal

--- On Mon, 12/1/08, Jan Kubr <jan.kubr at gmail.com> wrote:
From: Jan Kubr <jan.kubr at gmail.com>
Subject: Re: [Freeswitch-users] How to specify Path for sound files
To: freeswitch-users at lists.freeswitch.org
Date: Monday, December 1, 2008, 12:55 PM

Hi Faisal,
the path is either an absolute path or a path relative to the
directory in the sound_prefix var in vars.xml.
So this

<action application="playback"
data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"/>

works fine on my box. You sure this one doesn't work for you?

Jan

> I tried to play a sound file using the dialplan given on the link
> http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav
>
> <action application="playback"
data="/path/to/your.wav"/>
> In place of /path/to/your.wave I used
> "/en/us/callie/misc/8000/call_secured.wav"
>
"/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
> "/sounds/en/us/callie/misc/8000/call_secured.wav"
> But none of these is useful bcoz when i call on 2009, which is
> to b dialed to play the sound, same msg is
> displayed "404 NOT FOUND"
> Plz help me out.??? Faisal

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From hads at nice.net.nz  Mon Dec  1 21:54:01 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Tue, 2 Dec 2008 18:54:01 +1300
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <625821.37656.qm@web30706.mail.mud.yahoo.com>
References: <625821.37656.qm@web30706.mail.mud.yahoo.com>
Message-ID: <200812021854.01959.hads@nice.net.nz>

On Tuesday 02 December 2008 18:30:53 Faisal Maqsoodi wrote:
> Yes its not working on my system. When i copy this in default.xml dialplan,
> it works but as a seperate extension in dialplan/extensions it does'nt.
> Faisal

It's a little hard to understand what you're saying but I'd hazard a guess 
that your extension is below the transfer to enum.

Are you creating a separate file in conf/dialplan/default/ ? What are you 
naming the file? Does it show up below the enum file in a directory listing?

As Anthony said if you set debug logging then you will see what is going on.

hads
-- 
http://nicegear.co.nz
VoIP, DVB and other Linux compatible hardware.



From dave at 3c.co.uk  Mon Dec  1 22:11:35 2008
From: dave at 3c.co.uk (David Knell)
Date: Tue, 02 Dec 2008 06:11:35 +0000
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
Message-ID: <4934D197.8080007@3c.co.uk>

Hi Anthony,
> mod_openmrcp was a contribution to the community by a 3rd party 
> individual.
>
> As i have clearly stated in 2 previous emails, the man has decided to 
> discontinue the openmrcp project.
> So now we are left with the remains of the module and discontinued 
> code.  This was not our decision it was his.
I absolutely understand this but it's important, from a user point of 
view, to be able to know which bits of FS are current/supported and 
which aren't.
> Some people use it without issue which may mean that the crash you 
> reported is windows specific and I do not have a working lab of any 
> mrcp capbable system to try it against in unix for that matter.  I 
> have a list of work to do from here to the moon and back so on an 
> issue like this, unless someone can hand me login credentials to some 
> box and give me a phone number to dial to reporduce the issue, it will 
> be a long time until we can deal with it.  
It's useful to know that there are people using mod_openmrcp without 
issue: I did ask here if anyone was a while back, and no-one fessed up.  
I'll give it a go on a Linux box and report back.  And if you'd like a 
dev/test environment set up, then just tell me which one.
> And the question arises, should we bother working on it anymore if the 
> lib has been abandoned and we cannot even get any support from it's 
> author which is where the problem most likely lies.
>
> I try not to get too annoyed by these remarks about what we *ought to 
> do* because I know people lose sight of how much of the work to 
> support the project is done by a small group of 3 people and not the 
> 2000 people it appears to be from the outside looking in. (I've been 
> answering email for 4 hours now)
Those guys who claim to have all that money in an offshore bank account 
are lying - you don't have to reply to them in future ;-)  Seriously, 
though, I don't think it's too outrageous an idea to document what's 
supported and were you (for example) to have suggested that I get in 
touch with the contributors to the various modules, ask them what their 
view of its status is, condense the answers in to a list and report 
back, it's something I'd quite happily do.
> My suggestion is to pool some cash and pay the guy to make mod_unimrcp 
> for FS that we can maintain in tree knowing the development can be 
> supported by the original author.
Quite happy to participate in that, too.. the problem is that I've a 
demo to do like yesterday and the timescale for mod_unimrcp is a bit on 
the long side for that.  I'd rather not have to do it with Asterisk and 
Lumenvox..!

Cheers --

Dave
>
>
> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk 
> <mailto:dave at 3c.co.uk>> wrote:
>
>     Hi Mike,
>
>     My experience is that it's somewhat broken - it took two trivial
>     tweaks to get it to work with IBM's ASR and TTS, but there's a
>     more intractable problem to do with memory getting overwritten (I
>     assume that this is something to do with something being freed
>     when it shouldn't be) which causes a segfault on the second or
>     third session after the module being loaded.
>
>     Without wishing to sound like a stuck record, one thing that you
>     guys really ought to do is to decide what's supported and what
>     isn't, and make this obvious - for example, move unsupported
>     modules to a different place in the tree, don't have them built by
>     default, etc.  MRCP is in the specsheet on the Wiki.  Otherwise
>     folk like Mark and I spend time installing stuff, go round in
>     circles a bit trying to make it work, and then find out (a) that
>     it doesn't and (b) it's not going to be fixed because it's not
>     supported.
>
>     Cheers --
>
>     Dave
>>     I would not say it is totally broken, it is known to work in
>>     quite a few places, but we are unlikely to be doing any new fixes
>>     in it.
>>
>>     Mike
>>
>>     On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com
>>     <mailto:mszlazak at aol.com> wrote:
>>
>>>     Hi Anthony,
>>>
>>>     Oh! OK.
>>>
>>>     So is this module "totally broken".
>>>
>>>     I say this because I can't seem to get it to work at all with
>>>     the example in that Mod_openmrcp wiki page but I thought it
>>>     might because I'm not be using the right Cepstral software
>>>     (freetrial download versus the paided for SDK) or that I'm not
>>>     using the right port numbers or something else I didn't do. I
>>>     used TcpView to look at local port associated with my Cepstral
>>>     software and changed a few things but still nothing. I changed
>>>     the loglevel setting to 7 in the wiki's example but I don't see
>>>     the kind of output on the console that I would expect for debug
>>>     mode.
>>>
>>>     Thanks. Mark.
>>>
>>>
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     David Knell, Director, 3C Limited
>     T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>     http://www.3c.co.uk 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From yudha2008 at gmail.com  Mon Dec  1 22:24:06 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 2 Dec 2008 11:54:06 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
Message-ID: <d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>

*Hi Giovanni Maruzzelli*,

To list the available devices i have given this command *pa devlist*
*output:*
freeswitch at hp30094686650.optimus.co.in> pa devlist
2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process()
Unknown Command: pa

But when i check in my system *hwconf *there is auido drives

*class: AUDIO
bus: PCI
detached: 0
driver: snd-intel8x0
desc: "Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller"
vendorId: 8086
deviceId: 24d5
subVendorId: 8086
subDeviceId: 0c4a
pciType: 1
pcidom:    0
pcibus:  0
pcidev: 1f
pcifn:  5*

How to resolve the problem. Can u correct me where i am wrong.Can u just
describe what is the error also.

Thanks for reply
*--
Warm Regards,
N.Baskar
*
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From mszlazak at aol.com  Mon Dec  1 22:40:36 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Tue, 02 Dec 2008 01:40:36 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <8CB222E46498CA1-11E0-E8B@MBLK-M05.sysops.aol.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com><191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com><8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com><CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com><4934323B.9000305@3c.co.uk><191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<8CB222E46498CA1-11E0-E8B@MBLK-M05.sysops.aol.com>
Message-ID: <8CB226004412A8C-430-1FDA@WEBMAIL-MA12.sysops.aol.com>


 


 Just to follow up.

Moshe Yudkowsky has an article on "Routing calls from FreeSwitch to Prophecy":? http://www.prophecy2006.com/node/145

My problem is that Freeswitch and Prophecy need to be on the same machine BUT both need to bind to port 5060 so I'm getting errors from one or the other depending who's running first.

So can I change what port(s) FS uses and that way avoid this conflict? 
Maybe, this might let me bridge the call via FreeSwitch to Prophecy similar to what Moshe's article discusses???


 

-----Original Message-----
From: mszlazak at aol.com
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 1 Dec 2008 4:44 pm
Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP











 





 Does "bridging" a call from FS to Voxeo's Prophecy server require openMRCP? If not then the other issue I might have is a database look up that is part of the dialogue that maybe need as the person response to prompts from the asr. It's possible to run a php script for the database stuff that Prophecy might need or could that happen via Javascript in FS? Then after the dialogue has completed I go from Prophecy back to FS.





 



-----Original Message-----

From: Anthony Minessale <anthony.minessale at gmail.com>

To: freeswitch-users at lists.freeswitch.org

Sent: Mon, 1 Dec 2008 11:17 am

Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP












mod_openmrcp was a contribution to the community by a 3rd party individual.



As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project.

So now we are left with the remains of the module and discontinued code.? This was not our decision it was his. 




Since the author of openmrcp has stated that he has a new unimrcp we are certainly going to 

work towards getting mod_unimrcp to replace mod_openmrcp.? He had already commented on that previous thread to state he is willing to consider making a new module.





Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter.? I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it.? And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies.




I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) 




My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author.







On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:






  




Hi Mike,





My experience is that it's somewhat broken - it took two trivial tweaks
to get it to work with IBM's ASR and TTS, but there's a more
intractable problem to do with memory getting overwritten (I assume
that this is something to do with something being freed when it
shouldn't be) which causes a segfault on the second or third session
after the module being loaded.





Without wishing to sound like a stuck record, one thing that you guys
really ought to do is to decide what's supported and what isn't, and
make this obvious - for example, move unsupported modules to a
different place in the tree, don't have them built by default, etc.?
MRCP is in the specsheet on the Wiki.? Otherwise folk like Mark and I
spend time installing stuff, go round in circles a bit trying to make
it work, and then find out (a) that it doesn't and (b) it's not going
to be fixed because it's not supported.





Cheers --





Dave











I would not say it is totally broken, it is known to work
in quite a few places, but we are unlikely to be doing any new fixes in
it.
  




  


  

Mike


  




  


  

On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:


  


  

    

 Hi Anthony,


    


Oh! OK.


    


So is this module "totally broken". 


    


I say this because I can't seem to get it to work at all with the
example in that Mod_openmrcp wiki page but I thought it might because
I'm not be using the right Cepstral software (freetrial download versus
the paided for SDK) or that I'm not using the right port numbers or
something else I didn't do. I used TcpView to look at local port
associated with my Cepstral software and changed a few things but still
nothing. I changed the loglevel setting to 7 in the wiki's example but
I don't see the kind of output on the console that I would expect for
debug mode.


    


Thanks. Mark.


    


    

 


    


    




    


  

  


  


  










_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org

  












-- 

David Knell, Director, 3C Limited

T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031

http://www.3c.co.uk 







_______________________________________________


Freeswitch-users mailing list


Freeswitch-users at lists.freeswitch.org


http://lists.freeswitch.org/mailman/listinfo/freeswitch-users


UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users


http://www.freeswitch.org












-- 

Anthony Minessale II



FreeSWITCH http://www.freeswitch.org/

ClueCon http://www.cluecon.com/




AIM: anthm

MSN:anthony_minessale at hotmail.com

GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com


IRC: irc.freenode.net #freeswitch



FreeSWITCH Developer Conference

sip:888 at conference.freeswitch.org

iax:guest at conference.freeswitch.org/888


googletalk:conf+888 at conference.freeswitch.org

pstn:213-799-1400





 






_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





 





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_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From faisalmaqsoodi at yahoo.com  Mon Dec  1 22:44:21 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Mon, 1 Dec 2008 22:44:21 -0800 (PST)
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <200812021854.01959.hads@nice.net.nz>
Message-ID: <603330.68689.qm@web30704.mail.mud.yahoo.com>

Actually i copied the following text in a new text file and saved it as test1.xml file in /conf/dialplan/extensions, where 99999_enum.xml and 00_pizza_demo.xml exist, but it didnt worked.
<extension name="wavs">
    <condition field="destination_number" expression="^2009$">
      <action application="playback" data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"/>
    </condition>
  </extension>
Then i copied the same text and pasted in conf/dialplan/default.xml file below the line
<context name="default"> and above the line
<extension name="unloop">
and it worked successfully. Hope i ve explained what i wanted to.
??????????????????????????????????????????????????????????????????????????????????????? Faisal

--- On Mon, 12/1/08, Hadley Rich <hads at nice.net.nz> wrote:
From: Hadley Rich <hads at nice.net.nz>
Subject: Re: [Freeswitch-users] How to specify Path for sound files
To: freeswitch-users at lists.freeswitch.org
Date: Monday, December 1, 2008, 9:54 PM

On Tuesday 02 December 2008 18:30:53 Faisal Maqsoodi wrote:
> Yes its not working on my system. When i copy this in default.xml
dialplan,
> it works but as a seperate extension in dialplan/extensions it
does'nt.
> Faisal

It's a little hard to understand what you're saying but I'd hazard
a guess 
that your extension is below the transfer to enum.

Are you creating a separate file in conf/dialplan/default/ ? What are you 
naming the file? Does it show up below the enum file in a directory listing?

As Anthony said if you set debug logging then you will see what is going on.

hads
-- 
http://nicegear.co.nz
VoIP, DVB and other Linux compatible hardware.

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From gmaruzz at celliax.org  Mon Dec  1 22:50:56 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 2 Dec 2008 07:50:56 +0100
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
Message-ID: <7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>

Hi Baskar,

you have to compile and enable the module mod_portaudio.

Please edit the modules.conf in the main directory of the FS sources,
and remove the "#" before mod_portaudio. Also, after compilation and
installation ("make install"), in the directory
/usr/local/freeswitch/conf/autoload/ edit the file modules.conf.xml so
to enable the portaudio module and edit the portaudio.conf.xml to
reflect your setup.

Sincerely,

Giovanni Maruzzelli

=========================================
Cell : 39-347-2665618
Fax : 39-02-87390039



On Tue, Dec 2, 2008 at 7:24 AM, Baskar <yudha2008 at gmail.com> wrote:
> Hi Giovanni Maruzzelli,
>
> To list the available devices i have given this command pa devlist
> output:
> freeswitch at hp30094686650.optimus.co.in> pa devlist
> 2008-12-02 11:27:34 [CONSOLE] switch_console.c:255 switch_console_process()
> Unknown Command: pa
>
> But when i check in my system hwconf there is auido drives
>
> class: AUDIO
> bus: PCI
> detached: 0
> driver: snd-intel8x0
> desc: "Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller"
> vendorId: 8086
> deviceId: 24d5
> subVendorId: 8086
> subDeviceId: 0c4a
> pciType: 1
> pcidom:    0
> pcibus:  0
> pcidev: 1f
> pcifn:  5
>
> How to resolve the problem. Can u correct me where i am wrong.Can u just
> describe what is the error also.
>
> Thanks for reply
> --
> Warm Regards,
> N.Baskar
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mszlazak at aol.com  Mon Dec  1 23:00:32 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Tue, 02 Dec 2008 02:00:32 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
Message-ID: <8CB2262CD41B5B4-430-201D@WEBMAIL-MA12.sysops.aol.com>



 I need to barge in again and add to my last post with this email from Voxeo support. Here is their response to the port binding conflict and it brings up a possible problem if FreeSwitch will be looking for Prophecy at that port? I assumed it would if I set up the extension right but now I don't know and need your assistance with this issue ... as well. 

Thank you.

MESSAGE:
Hi Mark,

You are correct in that having multiple applications binding to the same port 
can cause a bundle of problems. You can configure Prophecy to stay away from 
port 5060, but then the question is whether FreeSwitch will be looking for 
Prophecy at that port (if its assuming that it's residing on a different box). 
Port 5060 is the standard for SIP traffic.

To get Prophecy off 5060 you will need to edit the config.xml and 
callrouting.xml files. You will need to search out all instances of "5060" and 
replace with, perhaps, port 5068. For instance:

<item name="ListenOnIP1">0.0.0.0:5068</item>
<item name="ListenOnIP2">0.0.0.0:5061</item>
<item name="ListenOnIP3">0.0.0.0:5067</item>
<item name="ListenOnIP4">0.0.0.0:5063</item>
<item name="ListenOnIP5">0.0.0.0:5064</item>
<item name="ListenOnIP6">0.0.0.0:5065</item>

instead of this...

<item name="ListenOnIP1">0.0.0.0:5060</item>
<item name="ListenOnIP2">0.0.0.0:5061</item>
<item name="ListenOnIP3">0.0.0.0:5062</item>
<item name="ListenOnIP4">0.0.0.0:5063</item>
<item name="ListenOnIP5">0.0.0.0:5064</item>
<item name="ListenOnIP6">0.0.0.0:5065</item>

Regards,
Jeff Kustermann
Voxeo Support




? 





 





 



-----Original Message-----

From: mszlazak at aol.com

To: freeswitch-users at lists.freeswitch.org

Sent: Mon, 1 Dec 2008 10:40 pm

Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP















 








 Just to follow up.





Moshe Yudkowsky has an article on "Routing calls from FreeSwitch to Prophecy":? http://www.prophecy2006.com/node/145





My problem is that Freeswitch and Prophecy need to be on the same machine BUT both need to bind to port 5060 so I'm getting errors from one or the other depending who's running first.





So can I change what port(s) FS uses and that way avoid this conflict? 


Maybe, this might let me bridge the call via FreeSwitch to Prophecy similar to what Moshe's article discusses???








 





-----Original Message-----


From: mszlazak at aol.com


To: freeswitch-users at lists.freeswitch.org


Sent: Mon, 1 Dec 2008 4:44 pm


Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP



















 











 Does "bridging" a call from FS to Voxeo's Prophecy server require openMRCP? If not then the other issue I might have is a database look up that is part of the dialogue that maybe need as the person response to prompts from the asr. It's possible to run a php script for the database stuff that Prophecy might need or could that happen via Javascript in FS? Then after the dialogue has completed I go from Prophecy back to FS.











 







-----Original Message-----



From: Anthony Minessale <anthony.minessale at gmail.com>



To: freeswitch-users at lists.freeswitch.org



Sent: Mon, 1 Dec 2008 11:17 am



Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP


















mod_openmrcp was a contribution to the community by a 3rd party individual.







As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project.



So now we are left with the remains of the module and discontinued code.? This was not our decision it was his. 








Since the author of openmrcp has stated that he has a new unimrcp we are certainly going to 



work towards getting mod_unimrcp to replace mod_openmrcp.? He had already commented on that previous thread to state he is willing to consider making a new module.









Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter.? I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it.? And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies.








I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) 








My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author.















On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:








  






Hi Mike,









My experience is that it's somewhat broken - it took two trivial tweaks
to get it to work with IBM's ASR and TTS, but there's a more
intractable problem to do with memory getting overwritten (I assume
that this is something to do with something being freed when it
shouldn't be) which causes a segfault on the second or third session
after the module being loaded.









Without wishing to sound like a stuck record, one thing that you guys
really ought to do is to decide what's supported and what isn't, and
make this obvious - for example, move unsupported modules to a
different place in the tree, don't have them built by default, etc.?
MRCP is in the specsheet on the Wiki.? Otherwise folk like Mark and I
spend time installing stuff, go round in circles a bit trying to make
it work, and then find out (a) that it doesn't and (b) it's not going
to be fixed because it's not supported.









Cheers --









Dave





















I would not say it is totally broken, it is known to work
in quite a few places, but we are unlikely to be doing any new fixes in
it.
  








  




  



Mike




  








  




  



On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:




  




  

    



 Hi Anthony,




    




Oh! OK.




    




So is this module "totally broken". 




    




I say this because I can't seem to get it to work at all with the
example in that Mod_openmrcp wiki page but I thought it might because
I'm not be using the right Cepstral software (freetrial download versus
the paided for SDK) or that I'm not using the right port numbers or
something else I didn't do. I used TcpView to look at local port
associated with my Cepstral software and changed a few things but still
nothing. I changed the loglevel setting to 7 in the wiki's example but
I don't see the kind of output on the console that I would expect for
debug mode.




    




Thanks. Mark.




    




    



 




    




    








    




  

  




  




  


























_______________________________________________







Freeswitch-users mailing list







Freeswitch-users at lists.freeswitch.org







http://lists.freeswitch.org/mailman/listinfo/freeswitch-users







UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users







http://www.freeswitch.org







  






















-- 







David Knell, Director, 3C Limited







T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031







http://www.3c.co.uk 











_______________________________________________




Freeswitch-users mailing list




Freeswitch-users at lists.freeswitch.org




http://lists.freeswitch.org/mailman/listinfo/freeswitch-users




UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users




http://www.freeswitch.org






















-- 



Anthony Minessale II







FreeSWITCH http://www.freeswitch.org/



ClueCon http://www.cluecon.com/








AIM: anthm



MSN:anthony_minessale at hotmail.com



GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com




IRC: irc.freenode.net #freeswitch







FreeSWITCH Developer Conference



sip:888 at conference.freeswitch.org



iax:guest at conference.freeswitch.org/888




googletalk:conf+888 at conference.freeswitch.org



pstn:213-799-1400









 








_______________________________________________







Freeswitch-users mailing list







Freeswitch-users at lists.freeswitch.org







http://lists.freeswitch.org/mailman/listinfo/freeswitch-users







UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users







http://www.freeswitch.org













 









 Tis the season to save your money!  Get the new AOL Holiday Toolbar for money saving offers and gift ideas. 







 







_______________________________________________



Freeswitch-users mailing list



Freeswitch-users at lists.freeswitch.org



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



http://www.freeswitch.org








 







 Tis the season to save your money!  Get the new AOL Holiday Toolbar for money saving offers and gift ideas. 





 






_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

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From hads at nice.net.nz  Mon Dec  1 23:03:35 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Tue, 2 Dec 2008 20:03:35 +1300
Subject: [Freeswitch-users] How to specify Path for sound files
In-Reply-To: <603330.68689.qm@web30704.mail.mud.yahoo.com>
References: <603330.68689.qm@web30704.mail.mud.yahoo.com>
Message-ID: <200812022003.36345.hads@nice.net.nz>

On Tuesday 02 December 2008 19:44:21 Faisal Maqsoodi wrote:
> Actually i copied the following text in a new text file and saved it as
> test1.xml file in /conf/dialplan/extensions, where 99999_enum.xml and
> 00_pizza_demo.xml exist, but it didnt worked. <extension name="wavs">
>     <condition field="destination_number" expression="^2009$">
>       <action application="playback"
> data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
>/> </condition>
>   </extension>
> Then i copied the same text and pasted in conf/dialplan/default.xml file
> below the line <context name="default"> and above the line
> <extension name="unloop">
> and it worked successfully. Hope i ve explained what i wanted to.
> ???????????????????????????????????????????????????????????????????????????
>???????????? Faisal
>
> --- On Mon, 12/1/08, Hadley Rich <hads at nice.net.nz> wrote:
> From: Hadley Rich <hads at nice.net.nz>
> Subject: Re: [Freeswitch-users] How to specify Path for sound files
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, December 1, 2008, 9:54 PM
>
> On Tuesday 02 December 2008 18:30:53 Faisal Maqsoodi wrote:
> > Yes its not working on my system. When i copy this in default.xml
>
> dialplan,
>
> > it works but as a seperate extension in dialplan/extensions it
>
> does'nt.
>
> > Faisal
>
> It's a little hard to understand what you're saying but I'd hazard
> a guess
> that your extension is below the transfer to enum.
>
> Are you creating a separate file in conf/dialplan/default/ ? What are you
> naming the file? Does it show up below the enum file in a directory
> listing?
>
> As Anthony said if you set debug logging then you will see what is going
> on.
>
> hads

People should really quote properly huh.

As I suspected you're getting transferred to enum (which the debug logging 
would have told you).

Try naming your file 50-test.xml and seeing what happens.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier



From faisalmaqsoodi at yahoo.com  Mon Dec  1 23:13:41 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Mon, 1 Dec 2008 23:13:41 -0800 (PST)
Subject: [Freeswitch-users] How to specify Path for sound files [DONE]
In-Reply-To: <200812022003.36345.hads@nice.net.nz>
Message-ID: <9442.5309.qm@web30706.mail.mud.yahoo.com>

Sir thank you very much. It really works.
?????????????????????????????????????????????????????????? Faisal

--- On Mon, 12/1/08, Hadley Rich <hads at nice.net.nz> wrote:
From: Hadley Rich <hads at nice.net.nz>
Subject: Re: [Freeswitch-users] How to specify Path for sound files
To: freeswitch-users at lists.freeswitch.org
Date: Monday, December 1, 2008, 11:03 PM

On Tuesday 02 December 2008 19:44:21 Faisal Maqsoodi wrote:
> Actually i copied the following text in a new text file and saved it as
> test1.xml file in /conf/dialplan/extensions, where 99999_enum.xml and
> 00_pizza_demo.xml exist, but it didnt worked. <extension
name="wavs">
>     <condition field="destination_number"
expression="^2009$">
>       <action application="playback"
>
data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"
>/> </condition>
>   </extension>
> Then i copied the same text and pasted in conf/dialplan/default.xml file
> below the line <context name="default"> and above the line
> <extension name="unloop">
> and it worked successfully. Hope i ve explained what i wanted to.
>
???????????????????????????????????????????????????????????????????????????
>???????????? Faisal
>
> --- On Mon, 12/1/08, Hadley Rich <hads at nice.net.nz> wrote:
> From: Hadley Rich <hads at nice.net.nz>
> Subject: Re: [Freeswitch-users] How to specify Path for sound files
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, December 1, 2008, 9:54 PM
>
> On Tuesday 02 December 2008 18:30:53 Faisal Maqsoodi wrote:
> > Yes its not working on my system. When i copy this in default.xml
>
> dialplan,
>
> > it works but as a seperate extension in dialplan/extensions it
>
> does'nt.
>
> > Faisal
>
> It's a little hard to understand what you're saying but I'd
hazard
> a guess
> that your extension is below the transfer to enum.
>
> Are you creating a separate file in conf/dialplan/default/ ? What are you
> naming the file? Does it show up below the enum file in a directory
> listing?
>
> As Anthony said if you set debug logging then you will see what is going
> on.
>
> hads

People should really quote properly huh.

As I suspected you're getting transferred to enum (which the debug logging 
would have told you).

Try naming your file 50-test.xml and seeing what happens.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From yudha2008 at gmail.com  Mon Dec  1 23:27:15 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 2 Dec 2008 12:57:15 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
Message-ID: <d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>

*Hi,

I have updated all the above events you told .It's working fine but when i
call extension 1002 from freeswitch console, call is connected to extension
1002, but FS is aborted but call is established in1002. what shall i do.
what was the error.

Full freeswitch get cut.*

*output:*
freeswitch at hp30094686650.optimus.co.in>* pa call 1002*
2008-12-02 12:54:05 [NOTICE] switch_channel.c:553 switch_channel_set_name()
New Channel portaudio/1002 [20b1163a-29c7-4369-bdb5-27398dc1a263]
2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call() Channel
[portaudio/1002] has been answered
API CALL [pa(call 1002)] output:
SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263

2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing
FreeSWITCH->1002 in context default
2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805 switch_ivr_set_user() can't
find user [default@]
freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]
mod_dptools.c:902 info_function() CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [portaudio/1002]
Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [L16]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [L16]
Channel-Write-Codec-Rate: [8000]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [FreeSWITCH]
Caller-Caller-ID-Number: [0000000000]
Caller-Network-Addr: [172.20.176.32]
Caller-Destination-Number: [1002]
Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
Caller-Source: [mod_portaudio]
Caller-Context: [default]
Caller-Channel-Name: [portaudio/1002]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1228202645898038]
Caller-Channel-Created-Time: [1228202645898038]
Caller-Channel-Answered-Time: [1228202647630133]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_channel_name: [portaudio/1002]
variable_endpoint_disposition: [ANSWER]
variable_read_codec: [L16]
variable_read_rate: [8000]
variable_write_codec: [L16]
variable_write_rate: [8000]
variable_use_profile: [nat]
variable_dialed_ext: [1002]
variable_current_application: [info]


2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer-State []n
2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML
features
2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2
record_session::/usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML
features
2008-12-02 12:54:07 [NOTICE] switch_channel.c:553 switch_channel_set_name()
New Channel sofia/internal/1002 at 172.20.179.201:23878;rinstance=de482996ac747c8d
[f7f80a05-be75-414b-bcea-4e5a34c3351e]
*freeswitch: src/switch_core_io.c:179: switch_core_session_read_frame:
Assertion `(*frame)->codec != ((void *)0)' failed.
Aborted (core dumped)
[root at hp30094686650 bin]#
*
*Thanks for the reply. Correct me were i am wrong.*

*Warm Regards,
N.Baskar
*
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From msc at freeswitch.org  Mon Dec  1 23:35:58 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 1 Dec 2008 23:35:58 -0800
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
Message-ID: <72441BC8-74C6-4490-B025-278E8C3F0CCF@freeswitch.org>

Does the core dump always happen in this call scenario? If so, can you  
get a back trace? Put it on pastebin. That will hopefully help narrow  
down the issue.

-MC

Sent from my iPhone

On Dec 1, 2008, at 11:27 PM, Baskar <yudha2008 at gmail.com> wrote:

> Hi,
>
> I have updated all the above events you told .It's working fine but  
> when i call extension 1002 from freeswitch console, call is  
> connected to extension 1002, but FS is aborted but call is  
> established in1002. what shall i do. what was the error.
>
> Full freeswitch get cut.
>
> output:
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-02 12:54:05 [NOTICE] switch_channel.c:553  
> switch_channel_set_name() New Channel portaudio/1002  
> [20b1163a-29c7-4369-bdb5-27398dc1a263]
> 2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call()  
> Channel [portaudio/1002] has been answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263
>
> 2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt()  
> Processing FreeSWITCH->1002 in context default
> 2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805  
> switch_ivr_set_user() can't find user [default@]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]  
> mod_dptools.c:902 info_function() CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228202645898038]
> Caller-Channel-Created-Time: [1228202645898038]
> Caller-Channel-Answered-Time: [1228202647630133]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer- 
> State []n
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1  
> execute_extension::dx XML features
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ 
> usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3  
> execute_extension::cf XML features
> 2008-12-02 12:54:07 [NOTICE] switch_channel.c:553  
> switch_channel_set_name() New Channel sofia/internal/1002 at 172.20.179.201 
> :23878;rinstance=de482996ac747c8d [f7f80a05-be75-414b- 
> bcea-4e5a34c3351e]
> freeswitch: src/switch_core_io.c:179:  
> switch_core_session_read_frame: Assertion `(*frame)->codec != ((void  
> *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
> Thanks for the reply. Correct me were i am wrong.
>
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gmaruzz at celliax.org  Mon Dec  1 23:39:38 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 2 Dec 2008 08:39:38 +0100
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
Message-ID: <7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>

Baskar,

that is bizarre.

Seems there is a problem with mod_sofia, the module that manages SIP
connection to the SIP client at 1002 extension.

Maybe someone else on the list can be of more help.

Sincerely,

Giovanni Maruzzelli

=========================================
Cell : 39-347-2665618
Fax : 39-02-87390039



On Tue, Dec 2, 2008 at 8:27 AM, Baskar <yudha2008 at gmail.com> wrote:
> Hi,
>
> I have updated all the above events you told .It's working fine but when i
> call extension 1002 from freeswitch console, call is connected to extension
> 1002, but FS is aborted but call is established in1002. what shall i do.
> what was the error.
>
> Full freeswitch get cut.
>
> output:
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-02 12:54:05 [NOTICE] switch_channel.c:553 switch_channel_set_name()
> New Channel portaudio/1002 [20b1163a-29c7-4369-bdb5-27398dc1a263]
> 2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call() Channel
> [portaudio/1002] has been answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263
>
> 2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing
> FreeSWITCH->1002 in context default
> 2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805 switch_ivr_set_user() can't
> find user [default@]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]
> mod_dptools.c:902 info_function() CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228202645898038]
> Caller-Channel-Created-Time: [1228202645898038]
> Caller-Channel-Answered-Time: [1228202647630133]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer-State []n
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML
> features
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2
> record_session::/usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML
> features
> 2008-12-02 12:54:07 [NOTICE] switch_channel.c:553 switch_channel_set_name()
> New Channel
> sofia/internal/1002 at 172.20.179.201:23878;rinstance=de482996ac747c8d
> [f7f80a05-be75-414b-bcea-4e5a34c3351e]
> freeswitch: src/switch_core_io.c:179: switch_core_session_read_frame:
> Assertion `(*frame)->codec != ((void *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
> Thanks for the reply. Correct me were i am wrong.
>
> Warm Regards,
> N.Baskar
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From r.pankratz at fh-wolfenbuettel.de  Tue Dec  2 01:12:13 2008
From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz)
Date: Tue, 02 Dec 2008 10:12:13 +0100
Subject: [Freeswitch-users] Dialing tone when placing a call with portaudio
In-Reply-To: <20081120154859.16192.qmail@server15.citromail.hu>
References: <20081120154859.16192.qmail@server15.citromail.hu>
Message-ID: <4934FBED.7030307@fh-wolfenbuettel.de>

Hello,
when using mod_portaudio for calling somebody I don't hear anything 
until the other party answers the call. Is it possible to play a dialing 
tone when the other party is ringing?

Best regards
   Ren? Pankratz




From mike at jerris.com  Tue Dec  2 02:49:40 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 05:49:40 -0500
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
Message-ID: <3F8D37CC-261F-4C5E-A9F3-64F9AD97F761@jerris.com>

What revision of freeswitch is this?  Can you please test this with  
svn trunk?

Mike

On Dec 2, 2008, at 2:27 AM, Baskar wrote:

> Hi,
>
> I have updated all the above events you told .It's working fine but  
> when i call extension 1002 from freeswitch console, call is  
> connected to extension 1002, but FS is aborted but call is  
> established in1002. what shall i do. what was the error.
>
> Full freeswitch get cut.
>
> output:
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-02 12:54:05 [NOTICE] switch_channel.c:553  
> switch_channel_set_name() New Channel portaudio/1002  
> [20b1163a-29c7-4369-bdb5-27398dc1a263]
> 2008-12-02 12:54:07 [NOTICE] mod_portaudio.c:1555 place_call()  
> Channel [portaudio/1002] has been answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:20b1163a-29c7-4369-bdb5-27398dc1a263
>
> 2008-12-02 12:54:07 [INFO] mod_dialplan_xml.c:232 dialplan_hunt()  
> Processing FreeSWITCH->1002 in context default
> 2008-12-02 12:54:07 [WARNING] switch_ivr.c:1805  
> switch_ivr_set_user() can't find user [default@]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-02 12:54:07 [INFO]  
> mod_dptools.c:902 info_function() CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [20b1163a-29c7-4369-bdb5-27398dc1a263]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228202645898038]
> Caller-Channel-Created-Time: [1228202645898038]
> Caller-Channel-Answered-Time: [1228202647630133]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-02 12:54:07 [INFO] mod_dptools.c:888 log_function() Answer- 
> State []n
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1  
> execute_extension::dx XML features
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ 
> usr/local/freeswitch/recordings/0000000000.2008-12-02-12-54-07.wav
> 2008-12-02 12:54:07 [INFO] switch_ivr_async.c:1536  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3  
> execute_extension::cf XML features
> 2008-12-02 12:54:07 [NOTICE] switch_channel.c:553  
> switch_channel_set_name() New Channel sofia/internal/1002 at 172.20.179.201 
> :23878;rinstance=de482996ac747c8d [f7f80a05-be75-414b- 
> bcea-4e5a34c3351e]
> freeswitch: src/switch_core_io.c:179:  
> switch_core_session_read_frame: Assertion `(*frame)->codec != ((void  
> *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
> Thanks for the reply. Correct me were i am wrong.
>
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Dec  2 02:51:59 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 05:51:59 -0500
Subject: [Freeswitch-users] Dialing tone when placing a call with
	portaudio
In-Reply-To: <4934FBED.7030307@fh-wolfenbuettel.de>
References: <20081120154859.16192.qmail@server15.citromail.hu>
	<4934FBED.7030307@fh-wolfenbuettel.de>
Message-ID: <ADE59C30-1DBE-4DB5-B96B-9EDA0A0DCE07@jerris.com>

What are you calling, sip I assume, this may be a case where the sip  
signaling is sending a 180 ringing instead of a 183 and we are not  
generating ringback in that case.  Can you please confirm that and  
test if setting the ringback channel variable before bridge fixes this  
issue?

Mike

On Dec 2, 2008, at 4:12 AM, Rene Pankratz wrote:

> Hello,
> when using mod_portaudio for calling somebody I don't hear anything
> until the other party answers the call. Is it possible to play a  
> dialing
> tone when the other party is ringing?




From woodydickson at gmail.com  Tue Dec  2 02:55:06 2008
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 2 Dec 2008 18:55:06 +0800
Subject: [Freeswitch-users] libfreeswitch question
Message-ID: <a44688a0812020255n2e1c7407ydb5145e025ce34e6@mail.gmail.com>

Hi,

I am just having a dumb question and hoping someone can help me.  I am
trying to run a c program with libfreeswitch embedded so I can use some
external mechanism to keep track of freeswitch, but I am having problem
while compiling:

[root at localhost fs]# gcc switchnode.c -I/usr/local/freeswitch/include
-L/usr/local/freeswitch/lib -lfreeswitch -lpthread
switchnode.c: In function 'main':
switchnode.c:11: warning: passing argument 1 of
'switch_core_init_and_modload' makes integer from pointer without a cast
switchnode.c:11: warning: passing argument 3 of
'switch_core_init_and_modload' from incompatible pointer type
/usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to
`clock_gettime'
/usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to
`uuid_generate'
/usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `crypt_r'
collect2: ld returned 1 exit status
[root at localhost fs]#


Does anyone know which library is missing?

Thanks,
Woody
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From yudha2008 at gmail.com  Tue Dec  2 02:57:27 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 2 Dec 2008 16:27:27 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
Message-ID: <d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>

Hi,

*After starting the freeswitch I try to dial a extension from console* *but
when i call extension 1002 from freeswitch console, call is connected to
extension 1002, but FS is aborted but call is established in1002. What shall
I do? What was the error?*
*
 I have pasted the console events in pastebin in this path:

**http://fr.pastebin.ca/1273382
**
What is the error? Can any one correct me where I am wrong and try to
resolve the problem.

I want to know why Fs Aborted what should be done to recover from Aborted.*


-- 
Warm Regards,
N.Baskar
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From mike at jerris.com  Tue Dec  2 03:00:25 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 06:00:25 -0500
Subject: [Freeswitch-users] libfreeswitch question
In-Reply-To: <a44688a0812020255n2e1c7407ydb5145e025ce34e6@mail.gmail.com>
References: <a44688a0812020255n2e1c7407ydb5145e025ce34e6@mail.gmail.com>
Message-ID: <0E13D02B-53C1-453C-A09C-5002749A9BB4@jerris.com>


On Dec 2, 2008, at 5:55 AM, Woody Dickson wrote:

> Hi,
>
> I am just having a dumb question and hoping someone can help me.  I  
> am trying to run a c program with libfreeswitch embedded so I can  
> use some external mechanism to keep track of freeswitch, but I am  
> having problem while compiling:
>
> [root at localhost fs]# gcc switchnode.c -I/usr/local/freeswitch/ 
> include -L/usr/local/freeswitch/lib -lfreeswitch -lpthread
> switchnode.c: In function 'main':
> switchnode.c:11: warning: passing argument 1 of  
> 'switch_core_init_and_modload' makes integer from pointer without a  
> cast
> switchnode.c:11: warning: passing argument 3 of  
> 'switch_core_init_and_modload' from incompatible pointer type

looks like you have the wrong var types you are passing here.

>
> /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to  
> `clock_gettime'

-lrt

>
> /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to  
> `uuid_generate'

-luuid

>
> /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to  
> `crypt_r'

-lcrypt

>
> collect2: ld returned 1 exit status
> [root at localhost fs]#
>
>
> Does anyone know which library is missing?




From mike at jerris.com  Tue Dec  2 03:02:05 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 06:02:05 -0500
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
Message-ID: <59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>

This appears to be a somewhat older version of svn trunk.  Please re- 
test with current svn trunk

Thanks
Mike

On Dec 2, 2008, at 5:57 AM, Baskar wrote:

> Hi,
>
> After starting the freeswitch I try to dial a extension from console  
> but when i call extension 1002 from freeswitch console, call is  
> connected to extension 1002, but FS is aborted but call is  
> established in1002. What shall I do? What was the error?
>
>  I have pasted the console events in pastebin in this path:
>
> http://fr.pastebin.ca/1273382
>
> What is the error? Can any one correct me where I am wrong and try  
> to resolve the problem.
>
> I want to know why Fs Aborted what should be done to recover from  
> Aborted.

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From yudha2008 at gmail.com  Tue Dec  2 03:07:20 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 2 Dec 2008 16:37:20 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<87f2f3b90811272355s7e5f1ab6l5d795bb62c0c50e0@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
Message-ID: <d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>

*Hi,

This is the svn version i have installed before a month

FreeSWITCH Version 1.0.trunk (10130M)
*
-- 
*Warm Regards,
N.Baskar*
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From keith.wood2000 at gmail.com  Tue Dec  2 03:23:14 2008
From: keith.wood2000 at gmail.com (Keith Wood)
Date: Tue, 2 Dec 2008 19:23:14 +0800
Subject: [Freeswitch-users] Problem with Freeswitch capturing DTMF
Message-ID: <b7737fe70812020323l23c93a17i2761b2947dc6b20b@mail.gmail.com>

Hi,

I am wondering if I am the only one getting this problem or not.  When
sending in DTMF to freeswitch, freeswitch is not always capable of capturing
all the DTMF being sent.  For instance, sending 1000 to freeswitch may end
up becoming 100 or 10003 becoming 1003.  Am I the only one getting this
strange issue?

If anyone know how to fix this problem, I would greatly appreciate it.

Regards,
Keith
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From mike at jerris.com  Tue Dec  2 03:45:28 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 06:45:28 -0500
Subject: [Freeswitch-users] Problem with Freeswitch capturing DTMF
In-Reply-To: <b7737fe70812020323l23c93a17i2761b2947dc6b20b@mail.gmail.com>
References: <b7737fe70812020323l23c93a17i2761b2947dc6b20b@mail.gmail.com>
Message-ID: <BBE9EE51-0FFD-400C-9AC1-4E93395D6080@jerris.com>

We generally are as good as possible on capturing dtmf reliably.  If  
you are seeing dropouts like that I would have to guess that this is a  
very lossy line.  Could you try and look at the packet capture of a  
call that is missing digits and see if you are indeed dropping a lot  
of packets.  If this is the case you could try info dtmf although that  
method has it's own issues.

Mike

On Dec 2, 2008, at 6:23 AM, "Keith Wood" <keith.wood2000 at gmail.com>  
wrote:

> Hi,
>
> I am wondering if I am the only one getting this problem or not.   
> When sending in DTMF to freeswitch, freeswitch is not always capable  
> of capturing all the DTMF being sent.  For instance, sending 1000 to  
> freeswitch may end up becoming 100 or 10003 becoming 1003.  Am I the  
> only one getting this strange issue?
>
> If anyone know how to fix this problem, I would greatly appreciate it.
>
> Regards,
> Keith
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dave at 3c.co.uk  Tue Dec  2 04:03:52 2008
From: dave at 3c.co.uk (David Knell)
Date: Tue, 02 Dec 2008 12:03:52 +0000
Subject: [Freeswitch-users] Problem with Freeswitch capturing DTMF
In-Reply-To: <BBE9EE51-0FFD-400C-9AC1-4E93395D6080@jerris.com>
References: <b7737fe70812020323l23c93a17i2761b2947dc6b20b@mail.gmail.com>
	<BBE9EE51-0FFD-400C-9AC1-4E93395D6080@jerris.com>
Message-ID: <49352428.4040706@3c.co.uk>

Hi Keith,

I was just writing a note along similar lines to Mike's.  If you need a 
hand getting
a packet capture or interpreting it, drop me a note off-list.

Cheers --

Dave

> We generally are as good as possible on capturing dtmf reliably.  If  
> you are seeing dropouts like that I would have to guess that this is a  
> very lossy line.  Could you try and look at the packet capture of a  
> call that is missing digits and see if you are indeed dropping a lot  
> of packets.  If this is the case you could try info dtmf although that  
> method has it's own issues.
>
> Mike
>
> On Dec 2, 2008, at 6:23 AM, "Keith Wood" <keith.wood2000 at gmail.com>  
> wrote:
>
>   
>> Hi,
>>
>> I am wondering if I am the only one getting this problem or not.   
>> When sending in DTMF to freeswitch, freeswitch is not always capable  
>> of capturing all the DTMF being sent.  For instance, sending 1000 to  
>> freeswitch may end up becoming 100 or 10003 becoming 1003.  Am I the  
>> only one getting this strange issue?
>>
>> If anyone know how to fix this problem, I would greatly appreciate it.
>>
>> Regards,
>> Keith
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From woodydickson at gmail.com  Tue Dec  2 04:05:33 2008
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 2 Dec 2008 20:05:33 +0800
Subject: [Freeswitch-users] Question about wrapping libfreeswitch
Message-ID: <a44688a0812020405k387df840k1ed1a33da370ab6f@mail.gmail.com>

Hi,

I am sorry again for sending another email to the group again.  I am working
on embedding libfreeswitch to provide better monitoring.  The first thing I
attempt to do is to run the sample  code provided in the website:

 #include <switch.h>
 int main(int argc, char **argv)
 {
   switch_core_flag_t flags = SCF_USE_SQL;
   int nc=0; /* this is for 'no console' mode, FALSE console is there, TRUE
it isnt */
   const char **err = NULL;     /* error value for return from freeswitch
initialization */
   #define LOGFILE "freeswitch.log"
   static char *lfile = LOGFILE; /* if NULL no logfile is generated */

   switch_core_init_and_modload(*lfile,flags,err);
   switch_core_runtime_loop(nc);
   switch_core_destroy();

   return (0); /* per C89 spec */
 }

But this code gives me segmentation fault when executing it.  This piece of
code is supposed to start up freeswitch and run it is a loop.  Does anyone
see what is wrong with it?  Does anyone have any working example that I can
refer to?

Thanks,
Woody
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From mike at jerris.com  Tue Dec  2 04:29:13 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 07:29:13 -0500
Subject: [Freeswitch-users] Question about wrapping libfreeswitch
In-Reply-To: <a44688a0812020405k387df840k1ed1a33da370ab6f@mail.gmail.com>
References: <a44688a0812020405k387df840k1ed1a33da370ab6f@mail.gmail.com>
Message-ID: <9719A132-2855-4D8D-BBE1-C64269E54C0D@jerris.com>

I think the api changed a little bit for this.  The easiest starting  
point would be to just clone switch.c and chop out any of the stuff  
you don't need, it's mostly argument handling code in there.

Mike

On Dec 2, 2008, at 7:05 AM, "Woody Dickson" <woodydickson at gmail.com>  
wrote:

> Hi,
>
> I am sorry again for sending another email to the group again.  I am  
> working on embedding libfreeswitch to provide better monitoring.   
> The first thing I attempt to do is to run the sample  code provided  
> in the website:
>
>  #include <switch.h>
>  int main(int argc, char **argv)
>  {
>    switch_core_flag_t flags = SCF_USE_SQL;
>    int nc=0; /* this is for 'no console' mode, FALSE console is  
> there, TRUE it isnt */
>    const char **err = NULL;     /* error value for return from  
> freeswitch  initialization */
>    #define LOGFILE "freeswitch.log"
>    static char *lfile = LOGFILE; /* if NULL no logfile is generated */
>
>    switch_core_init_and_modload(*lfile,flags,err);
>    switch_core_runtime_loop(nc);
>    switch_core_destroy();
>
>    return (0); /* per C89 spec */
>  }
>
> But this code gives me segmentation fault when executing it.  This  
> piece of code is supposed to start up freeswitch and run it is a  
> loop.  Does anyone see what is wrong with it?  Does anyone have any  
> working example that I can refer to?
>
> Thanks,
> Woody
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From c_cav_01 at yahoo.com  Tue Dec  2 05:25:07 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 05:25:07 -0800 (PST)
Subject: [Freeswitch-users]  Wrong # in voicemail
Message-ID: <20791453.post@talk.nabble.com>


My dialplan is pretty simple.  I have a single trunk with a vonage softphone
DID (1303... we'll call it main) and a "virtual" DID (1816...) which rings
the softphone DID.  All incoming calls show up as from softphone DID but the
sip_to_user holds the actual number dialed so I can enter the dialplan
properly.

I have 2 extensions in my directory/extensions, one for each of the DID's. 
The extensions check sip_to_user for match and that works great.  I match on
([0,1]?)(<10 digit did>) and it enters the dialplans correctly, plays the
right music for each DID while the dial is occuring, so all that works.  The
bridge to user/$2@$${domain} also works fine.

The continue_on_fail is set properly so on no answer call_timeout hits (at
25 secs), and goes to voicemail...  works also for both numbers.

transfer to voicemail is as follows
 <action application="answer"/>
 <action application="voicemail" data="default $${domain} $2"/>

which should be pulling $2 from the condition check shown above, which it
does, cuz the bridge works...

When I call in on main DID, I get "leave a message for 1303..."  The Main
DID..
When I call in on virtual, I get "leave a message for 1303..."  The Main DID
rather than the 1816....

How can I get voicemail to use the correct DID.    HELP!!  :D
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20791453.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From odermann at googlemail.com  Tue Dec  2 05:48:15 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 14:48:15 +0100
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
Message-ID: <5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>

we configured mod_shout and are able to record mp3. but if we start to
playback the file, it will only be played back to that point, which
was recorded, when we started the player.
we do this with "api uuid_record uuid start /var/www/test.mp3".

we are also able to playback a (radio-)stream to an uuid with
shout://ip-adress:12345

but what do we have to do, to listen to the file/stream with a player?

it seems, that fs has to stream to recording file to a streaming
server (like icecast), right? but if we do "api uuid_record uuid start
shout://user:passwd at ip-adress:12345/" (and other combinations), we get
an error:
2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid URL: xxxxx
2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851
switch_ivr_record_session() Error opening shout://xxxx

are we on the right track? is there something else we have to do to
make it work?


thanks for your help.



2008/12/1 Anthony Minessale <anthony.minessale at gmail.com>:
> yes,
>
> mod_shout will broadcast calls as MP3 that you can listen to in
> itunes/winamp live.



From brian at freeswitch.org  Tue Dec  2 06:45:47 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 08:45:47 -0600
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20791453.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com>
Message-ID: <B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>

Can you show me the full XML for this extension including the regular  
expression?

/b

On Dec 2, 2008, at 7:25 AM, ccav wrote:

> transfer to voicemail is as follows
> <action application="answer"/>
> <action application="voicemail" data="default $${domain} $2"/>




From brian at freeswitch.org  Tue Dec  2 06:47:18 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 08:47:18 -0600
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
Message-ID: <375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>

Are you on SVN trunk or what rev are you trying to use?

/b

On Dec 2, 2008, at 7:48 AM, Dennis wrote:

> it seems, that fs has to stream to recording file to a streaming
> server (like icecast), right? but if we do "api uuid_record uuid start
> shout://user:passwd at ip-adress:12345/" (and other combinations), we get
> an error:
> 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid  
> URL: xxxxx
> 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851
> switch_ivr_record_session() Error opening shout://xxxx




From odermann at googlemail.com  Tue Dec  2 07:03:24 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 16:03:24 +0100
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
Message-ID: <5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>

i am using the latest svn trunk from today.


2008/12/2 Brian West <brian at freeswitch.org>:
> Are you on SVN trunk or what rev are you trying to use?
>
> /b
>
> On Dec 2, 2008, at 7:48 AM, Dennis wrote:
>
>> it seems, that fs has to stream to recording file to a streaming
>> server (like icecast), right? but if we do "api uuid_record uuid start
>> shout://user:passwd at ip-adress:12345/" (and other combinations), we get
>> an error:
>> 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid
>> URL: xxxxx
>> 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851
>> switch_ivr_record_session() Error opening shout://xxxx
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Dec  2 07:08:30 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 09:08:30 -0600
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
	<5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
Message-ID: <5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>

And you have your shoutcast/icecast server set up and functional?

/b

On Dec 2, 2008, at 9:03 AM, Dennis wrote:

> i am using the latest svn trunk from today.




From odermann at googlemail.com  Tue Dec  2 07:25:20 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 16:25:20 +0100
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
	<5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
	<5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
Message-ID: <5e414ed0812020725y138737beu54cc5d0808161093@mail.gmail.com>

no, not yet. i am still fiddling arround with icecast2.

we tried it with someone, who offers radiostreams. perhaps this just
works with icecast(2) and shoutcast?



2008/12/2 Brian West <brian at freeswitch.org>:
> And you have your shoutcast/icecast server set up and functional?
>
> /b
>
> On Dec 2, 2008, at 9:03 AM, Dennis wrote:
>
>> i am using the latest svn trunk from today.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Dec  2 07:34:39 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 09:34:39 -0600
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5e414ed0812020725y138737beu54cc5d0808161093@mail.gmail.com>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
	<5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
	<5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
	<5e414ed0812020725y138737beu54cc5d0808161093@mail.gmail.com>
Message-ID: <A6D64AAE-DB9B-4206-B76B-81FE07F490AD@freeswitch.org>

icecast2 is a known working server we have talked to before.

/b

On Dec 2, 2008, at 9:25 AM, Dennis wrote:

> no, not yet. i am still fiddling arround with icecast2.
>
> we tried it with someone, who offers radiostreams. perhaps this just
> works with icecast(2) and shoutcast?




From anthony.minessale at gmail.com  Tue Dec  2 07:51:39 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 09:51:39 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <4934D197.8080007@3c.co.uk>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
Message-ID: <191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>

If you can get it to break on linux I will ssh in and fix it for you.
If you cannot, i can try to fix it for you over rdp but that won't be very
fun.

We can think about reinstating mod_lumenvox as well as another windows based
asr
alternative.  I deleted it for the same reason we will probably delete
mod_openmrcp because
nobody was using it and there was no way to support it because our dev
licenses had expired.

Lumenvox has offered us some new dev licenses to bring it back but I would
need someone to actually want it to work to put in charge of it.

We will be clear about what is supported and what is not in the 1.0.2
release scheduled
to be released in the near future.




On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:

>  Hi Anthony,
>
> mod_openmrcp was a contribution to the community by a 3rd party individual.
>
> As i have clearly stated in 2 previous emails, the man has decided to
> discontinue the openmrcp project.
> So now we are left with the remains of the module and discontinued code.
> This was not our decision it was his.
>
> I absolutely understand this but it's important, from a user point of view,
> to be able to know which bits of FS are current/supported and which aren't.
>
> Some people use it without issue which may mean that the crash you reported
> is windows specific and I do not have a working lab of any mrcp capbable
> system to try it against in unix for that matter.  I have a list of work to
> do from here to the moon and back so on an issue like this, unless someone
> can hand me login credentials to some box and give me a phone number to dial
> to reporduce the issue, it will be a long time until we can deal with it.
>
> It's useful to know that there are people using mod_openmrcp without issue:
> I did ask here if anyone was a while back, and no-one fessed up.  I'll give
> it a go on a Linux box and report back.  And if you'd like a dev/test
> environment set up, then just tell me which one.
>
> And the question arises, should we bother working on it anymore if the lib
> has been abandoned and we cannot even get any support from it's author which
> is where the problem most likely lies.
>
> I try not to get too annoyed by these remarks about what we *ought to do*
> because I know people lose sight of how much of the work to support the
> project is done by a small group of 3 people and not the 2000 people it
> appears to be from the outside looking in. (I've been answering email for 4
> hours now)
>
> Those guys who claim to have all that money in an offshore bank account are
> lying - you don't have to reply to them in future ;-)  Seriously, though, I
> don't think it's too outrageous an idea to document what's supported and
> were you (for example) to have suggested that I get in touch with the
> contributors to the various modules, ask them what their view of its status
> is, condense the answers in to a list and report back, it's something I'd
> quite happily do.
>
> My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
> FS that we can maintain in tree knowing the development can be supported by
> the original author.
>
> Quite happy to participate in that, too.. the problem is that I've a demo
> to do like yesterday and the timescale for mod_unimrcp is a bit on the long
> side for that.  I'd rather not have to do it with Asterisk and Lumenvox..!
>
> Cheers --
>
> Dave
>
>
>
> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>
>>  Hi Mike,
>>
>> My experience is that it's somewhat broken - it took two trivial tweaks to
>> get it to work with IBM's ASR and TTS, but there's a more intractable
>> problem to do with memory getting overwritten (I assume that this is
>> something to do with something being freed when it shouldn't be) which
>> causes a segfault on the second or third session after the module being
>> loaded.
>>
>> Without wishing to sound like a stuck record, one thing that you guys
>> really ought to do is to decide what's supported and what isn't, and make
>> this obvious - for example, move unsupported modules to a different place in
>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>> round in circles a bit trying to make it work, and then find out (a) that it
>> doesn't and (b) it's not going to be fixed because it's not supported.
>>
>> Cheers --
>>
>> Dave
>>
>>  I would not say it is totally broken, it is known to work in quite a few
>> places, but we are unlikely to be doing any new fixes in it.
>>  Mike
>>
>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>
>>  Hi Anthony,
>>
>> Oh! OK.
>>
>> So is this module "totally broken".
>>
>> I say this because I can't seem to get it to work at all with the example
>> in that Mod_openmrcp wiki page but I thought it might because I'm not be
>> using the right Cepstral software (freetrial download versus the paided for
>> SDK) or that I'm not using the right port numbers or something else I didn't
>> do. I used TcpView to look at local port associated with my Cepstral
>> software and changed a few things but still nothing. I changed the loglevel
>> setting to 7 in the wiki's example but I don't see the kind of output on the
>> console that I would expect for debug mode.
>>
>> Thanks. Mark.
>>
>>
>>    ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> David Knell, Director, 3C Limited
> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Tue Dec  2 07:54:57 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 09:54:57 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <8CB2262CD41B5B4-430-201D@WEBMAIL-MA12.sysops.aol.com>
References: <8CB2262CD41B5B4-430-201D@WEBMAIL-MA12.sysops.aol.com>
Message-ID: <191c3a030812020754t26951be6i904fcf19f38a02d9@mail.gmail.com>

FreeSWITCH has an enterprise scale SIP UA.  Not only can it listen on other
ports it can
listen and work on as many ip:port combos as you want simultaneously each
with it's own specific config.

If you have an affinity for port 5060 you can always bring up 2 IP on the
same box and give one to each application.  You can essentially do whatever
you want. It's your box and everything involved is configurable.




On Tue, Dec 2, 2008 at 1:00 AM, <mszlazak at aol.com> wrote:

>  I need to barge in again and add to my last post with this email from
> Voxeo support. Here is their response to the port binding conflict and it
> brings up a possible problem if FreeSwitch will be looking for Prophecy at
> that port? I assumed it would if I set up the extension right but now I
> don't know and need your assistance with this issue ... as well.
>
> Thank you.
>
> MESSAGE:
>
> Hi Mark,
>
>
> You are correct in that having multiple applications binding to the same port
>
> can cause a bundle of problems. You can configure Prophecy to stay away from
>
> port 5060, but then the question is whether FreeSwitch will be looking for
>
> Prophecy at that port (if its assuming that it's residing on a different box).
>
> Port 5060 is the standard for SIP traffic.
>
>
> To get Prophecy off 5060 you will need to edit the config.xml and
>
> callrouting.xml files. You will need to search out all instances of "5060" and
>
> replace with, perhaps, port 5068. For instance:
>
>
> <item name="ListenOnIP1">0.0.0.0:5068</item>
>
> <item name="ListenOnIP2">0.0.0.0:5061</item>
>
> <item name="ListenOnIP3">0.0.0.0:5067</item>
>
> <item name="ListenOnIP4">0.0.0.0:5063</item>
>
> <item name="ListenOnIP5">0.0.0.0:5064</item>
>
> <item name="ListenOnIP6">0.0.0.0:5065</item>
>
>
> instead of this...
>
>
> <item name="ListenOnIP1">0.0.0.0:5060</item>
>
> <item name="ListenOnIP2">0.0.0.0:5061</item>
>
> <item name="ListenOnIP3">0.0.0.0:5062</item>
>
> <item name="ListenOnIP4">0.0.0.0:5063</item>
>
> <item name="ListenOnIP5">0.0.0.0:5064</item>
>
> <item name="ListenOnIP6">0.0.0.0:5065</item>
>
>
> Regards,
>
> Jeff Kustermann
>
> Voxeo Support
>
>
>
>
>
>  -----Original Message-----
> From: mszlazak at aol.com
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 1 Dec 2008 10:40 pm
> Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP
>
>
>  Just to follow up.
>
> Moshe Yudkowsky has an article on "Routing calls from FreeSwitch to
> Prophecy":  http://www.prophecy2006.com/node/145
>
> My problem is that Freeswitch and Prophecy need to be on the same machine
> BUT both need to bind to port 5060 so I'm getting errors from one or the
> other depending who's running first.
>
> So can I change what port(s) FS uses and that way avoid this conflict?
> Maybe, this might let me bridge the call via FreeSwitch to Prophecy similar
> to what Moshe's article discusses???
>
>  -----Original Message-----
> From: mszlazak at aol.com
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 1 Dec 2008 4:44 pm
> Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP
>
>
>  Does "bridging" a call from FS to Voxeo's Prophecy server require
> openMRCP? If not then the other issue I might have is a database look up
> that is part of the dialogue that maybe need as the person response to
> prompts from the asr. It's possible to run a php script for the database
> stuff that Prophecy might need or could that happen via Javascript in FS?
> Then after the dialogue has completed I go from Prophecy back to FS.
>
>  -----Original Message-----
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 1 Dec 2008 11:17 am
> Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP
>
>  mod_openmrcp was a contribution to the community by a 3rd party
> individual.
>
> As i have clearly stated in 2 previous emails, the man has decided to
> discontinue the openmrcp project.
> So now we are left with the remains of the module and discontinued code.
> This was not our decision it was his.
>
> Since the author of openmrcp has stated that he has a new unimrcp we are
> certainly going to
> work towards getting mod_unimrcp to replace mod_openmrcp.  He had already
> commented on that previous thread to state he is willing to consider making
> a new module.
>
> Some people use it without issue which may mean that the crash you reported
> is windows specific and I do not have a working lab of any mrcp capbable
> system to try it against in unix for that matter.  I have a list of work to
> do from here to the moon and back so on an issue like this, unless someone
> can hand me login credentials to some box and give me a phone number to dial
> to reporduce the issue, it will be a long time until we can deal with it.
> And the question arises, should we bother working on it anymore if the lib
> has been abandoned and we cannot even get any support from it's author which
> is where the problem most likely lies.
>
> I try not to get too annoyed by these remarks about what we *ought to do*
> because I know people lose sight of how much of the work to support the
> project is done by a small group of 3 people and not the 2000 people it
> appears to be from the outside looking in. (I've been answering email for 4
> hours now)
>
> My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
> FS that we can maintain in tree knowing the development can be supported by
> the original author.
>
>
> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>
>>  Hi Mike,
>>
>> My experience is that it's somewhat broken - it took two trivial tweaks to
>> get it to work with IBM's ASR and TTS, but there's a more intractable
>> problem to do with memory getting overwritten (I assume that this is
>> something to do with something being freed when it shouldn't be) which
>> causes a segfault on the second or third session after the module being
>> loaded.
>>
>> Without wishing to sound like a stuck record, one thing that you guys
>> really ought to do is to decide what's supported and what isn't, and make
>> this obvious - for example, move unsupported modules to a different place in
>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>> round in circles a bit trying to make it work, and then find out (a) that it
>> doesn't and (b) it's not going to be fixed because it's not supported.
>>
>> Cheers --
>>
>> Dave
>>
>>  I would not say it is totally broken, it is known to work in quite a few
>> places, but we are unlikely to be doing any new fixes in it.
>>  Mike
>>
>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>
>>  Hi Anthony,
>>
>> Oh! OK.
>>
>> So is this module "totally broken".
>>
>> I say this because I can't seem to get it to work at all with the example
>> in that Mod_openmrcp wiki page but I thought it might because I'm not be
>> using the right Cepstral software (freetrial download versus the paided for
>> SDK) or that I'm not using the right port numbers or something else I didn't
>> do. I used TcpView to look at local port associated with my Cepstral
>> software and changed a few things but still nothing. I changed the loglevel
>> setting to 7 in the wiki's example but I don't see the kind of output on the
>> console that I would expect for debug mode.
>>
>> Thanks. Mark.
>>
>>
>>    ------------------------------
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> --
>>
>>
>>
>>
>>
>>
>>
>>
>> David Knell, Director, 3C Limited
>>
>>
>>
>>
>>
>>
>>
>>
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>>
>>
>>
>>
>>
>>
>>
>> http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
>
>
>
>
>
>
>
>
> Freeswitch-users mailing list
>
>
>
>
>
>
>
> Freeswitch-users at lists.freeswitch.org
>
>
>
>
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
>
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
>
>
>
>
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
> ------------------------------
> Tis the season to save your money! Get the new AOL Holiday Toolbar<http://toolbar.aol.com/holiday/download.html?ncid=emlweusdown00000008>for money saving offers and gift ideas.
>
> _______________________________________________
>
>
>
>
> Freeswitch-users mailing list
>
>
>
> Freeswitch-users at lists.freeswitch.org
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
> http://www.freeswitch.org
>
>
>
>
> ------------------------------
> Tis the season to save your money! Get the new AOL Holiday Toolbar<http://toolbar.aol.com/holiday/download.html?ncid=emlweusdown00000008>for money saving offers and gift ideas.
>
> _______________________________________________
>
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> ------------------------------
> Tis the season to save your money! Get the new AOL Holiday Toolbar<http://toolbar.aol.com/holiday/download.html?ncid=emlweusdown00000008>for money saving offers and gift ideas.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Tue Dec  2 08:07:21 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 17:07:21 +0100
Subject: [Freeswitch-users] Listen to a file, while recording?
In-Reply-To: <5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
References: <5e414ed0811180428s41db3533r3e61a59dd9a8bcf9@mail.gmail.com>
	<191c3a030811180756l1900f2e5x2ae59e6587abcb95@mail.gmail.com>
	<5e414ed0811280651l2801702flc694a5cd7761dea2@mail.gmail.com>
	<191c3a030812010929q72180f60ib3de60e5c91dc03@mail.gmail.com>
	<5e414ed0812020548k61d3b361wd56d9ae3cb4d1050@mail.gmail.com>
	<375BB16E-2DB0-4D14-811E-5C91FF4FFC99@freeswitch.org>
	<5e414ed0812020703w32c19eapeaf3122615c80207@mail.gmail.com>
	<5C8C992C-37D5-4358-8B97-F1BE9F86D6D8@freeswitch.org>
Message-ID: <5e414ed0812020807n77f4b3d3pd9cd18d72e416029@mail.gmail.com>

sorry, problem solved :-)

it works very good with icecast2.



2008/12/2 Brian West <brian at freeswitch.org>:
> And you have your shoutcast/icecast server set up and functional?
>
> /b
>
> On Dec 2, 2008, at 9:03 AM, Dennis wrote:
>
>> i am using the latest svn trunk from today.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Tue Dec  2 08:09:09 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 10:09:09 -0600
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<d0640d380812010254t576ec088nea7e68ec418bbd6d@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
Message-ID: <191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>

from the source tree of FS please type

"make current"

when it completes, retest the call.



On Tue, Dec 2, 2008 at 5:07 AM, Baskar <yudha2008 at gmail.com> wrote:

> *Hi,
>
> This is the svn version i have installed before a month
>
> FreeSWITCH Version 1.0.trunk (10130M)
> *
> --
> *Warm Regards,
> N.Baskar*
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From carlos.talbot at gmail.com  Tue Dec  2 08:13:28 2008
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 2 Dec 2008 10:13:28 -0600
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <000001c9530d$912d86d0$b3889470$@com>
References: <000001c9530d$912d86d0$b3889470$@com>
Message-ID: <5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>

Have you tried the latest msi build? It's based off svn 10564.

Carlos

On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:

> I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
> on a Windows Vista machine using the precompiled .msi - actually the same
> machine).
>
> Using the default configuration files, and using 2 Snom 360 phones I dialed
> from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
> instantly, but under Windows it takes 1-2 seconds before it starts ringing.
>
> It seems to be in the dialplan the time is spent. From the time I see this
> line on the console:
>
> [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
> context default
>
> Until the next thing happens it always takes at least 1 full second, but on
> the Mac it happens instantly.
>
> Why is the Windows build this much slower? Is it a known problem?
>
> I get the feeling that the majority of the FS community is Unix based,
> which
> is fine by me, but I would really like to know just how well supported and
> stable the Win32 build is and if this is currently a viable way to go, or
> if
> I should stick to Linux/BSD/Mac for production use?
>
>
> // Per
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stkn at freeswitch.org  Tue Dec  2 08:20:16 2008
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Tue, 2 Dec 2008 17:20:16 +0100
Subject: [Freeswitch-users] Support for Junghanns duoBRI
In-Reply-To: <B22D2E3D-9C6F-43C6-A634-DC00B91BDBBB@jerris.com>
References: <49340242.3040403@gmail.com>
	<B22D2E3D-9C6F-43C6-A634-DC00B91BDBBB@jerris.com>
Message-ID: <200812021720.16188.stkn@freeswitch.org>

All HFC-based cards supported by bristuffed Zaptel should work.

Stefan

Am Monday 01 December 2008 schrieb Michael Jerris:
> The bri support is still in development, basic calls on ptmp bri do  
> appear to work, although I am not sure with what hardware.
> 
> Mike
> 
> 
> On Dec 1, 2008, at 10:26 AM, Sergey Kirillov wrote:
> 
> > Greetings,
> >
> > Can somebody tell me, if it is possible to use duoBRI card
> > (http://www.junghanns.net/en/duobri_express_produkt.html) from
> > Junghanns.net together with Freeswitch?
> >
> > I've found that this card has Zaptel drivers, and Freeswitch has
> > mod_openzap. On the other side, I saw somewhere in wiki that  
> > Freeswitch
> > does not support BRI at all at the moment.
> >
> >
> > Please confirm or allay my apprehensions.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


-- 
Stefan Knoblich
Systemadministrator

axsentis GmbH
Eupener Strasse 74
50933 K?ln

Tel: 0180 - 506 705 521*
Fax: 0180 - 506 705 529*

E-Mail: s.knoblich at axsentis.de
Web: www.axsentis.de

Eingetragen beim AG K?ln: HR B 56238
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Gesellschafter-Gesch?ftsf?hrer: 
Yan Lecomte, Eduard Schlein, Apostolos Varsamis

*14ct/min aus dem Festnetz der T-Com | dtms



From gilbertandrew at me.com  Tue Dec  2 08:24:48 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 02 Dec 2008 11:24:48 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
	<191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
Message-ID: <39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>

Mark and David,

I am willing to help some with testing here as well, if you need it.  
Ping me directly or we can get on the IRC. I am on Mac OS, but have  
readily available vm's with Debian, etc. I also have Prophecy.

I have a general interest in an ASR solution as well. Voxeo is great,  
but using it as an MRCP proxy seems odd. As a full fledged VXML  
solution it is great, if you can afford it. But having a good ASR  
solution is good first step to trying to get something like OpenVXI  
working as well.

That said, seems like a bounty or money to help FS is a better spend  
anyway. It is a one time cost, not a variable cost. And it goes  
straight to the guys doing the real work.

I built unimrcp last night, it was quite straight forward. In theory,  
if I weren't old and my C/autoconf skills rather atrophied, it  
wouldn't seem like it would be that huge a deal to port/fix openmrcp  
to unimrcp.

Finally, Anthony I was looking at the Lumenvox path as well, but got  
deterred by the licensing hassle. This seems to be a universal ASR  
issue. I would reason I can find the old module in SVN? Were they  
going to grant "community dev" licenses? Again - I am willing to  
volunteer to do some testing/doc at least.

Andy



On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote:

> If you can get it to break on linux I will ssh in and fix it for you.
> If you cannot, i can try to fix it for you over rdp but that won't  
> be very fun.
>
> We can think about reinstating mod_lumenvox as well as another  
> windows based asr
> alternative.  I deleted it for the same reason we will probably  
> delete mod_openmrcp because
> nobody was using it and there was no way to support it because our  
> dev licenses had expired.
>
> Lumenvox has offered us some new dev licenses to bring it back but I  
> would need someone to actually want it to work to put in charge of it.
>
> We will be clear about what is supported and what is not in the  
> 1.0.2 release scheduled
> to be released in the near future.
>
>
>
>
> On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:
> Hi Anthony,
>
>> mod_openmrcp was a contribution to the community by a 3rd party  
>> individual.
>>
>> As i have clearly stated in 2 previous emails, the man has decided  
>> to discontinue the openmrcp project.
>> So now we are left with the remains of the module and discontinued  
>> code.  This was not our decision it was his.
> I absolutely understand this but it's important, from a user point  
> of view, to be able to know which bits of FS are current/supported  
> and which aren't.
>
>> Some people use it without issue which may mean that the crash you  
>> reported is windows specific and I do not have a working lab of any  
>> mrcp capbable system to try it against in unix for that matter.  I  
>> have a list of work to do from here to the moon and back so on an  
>> issue like this, unless someone can hand me login credentials to  
>> some box and give me a phone number to dial to reporduce the issue,  
>> it will be a long time until we can deal with it.
> It's useful to know that there are people using mod_openmrcp without  
> issue: I did ask here if anyone was a while back, and no-one fessed  
> up.  I'll give it a go on a Linux box and report back.  And if you'd  
> like a dev/test environment set up, then just tell me which one.
>
>> And the question arises, should we bother working on it anymore if  
>> the lib has been abandoned and we cannot even get any support from  
>> it's author which is where the problem most likely lies.
>>
>> I try not to get too annoyed by these remarks about what we *ought  
>> to do* because I know people lose sight of how much of the work to  
>> support the project is done by a small group of 3 people and not  
>> the 2000 people it appears to be from the outside looking in. (I've  
>> been answering email for 4 hours now)
> Those guys who claim to have all that money in an offshore bank  
> account are lying - you don't have to reply to them in future ;-)   
> Seriously, though, I don't think it's too outrageous an idea to  
> document what's supported and were you (for example) to have  
> suggested that I get in touch with the contributors to the various  
> modules, ask them what their view of its status is, condense the  
> answers in to a list and report back, it's something I'd quite  
> happily do.
>
>> My suggestion is to pool some cash and pay the guy to make  
>> mod_unimrcp for FS that we can maintain in tree knowing the  
>> development can be supported by the original author.
> Quite happy to participate in that, too.. the problem is that I've a  
> demo to do like yesterday and the timescale for mod_unimrcp is a bit  
> on the long side for that.  I'd rather not have to do it with  
> Asterisk and Lumenvox..!
>
> Cheers --
>
> Dave
>
>>
>>
>> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>> Hi Mike,
>>
>> My experience is that it's somewhat broken - it took two trivial  
>> tweaks to get it to work with IBM's ASR and TTS, but there's a more  
>> intractable problem to do with memory getting overwritten (I assume  
>> that this is something to do with something being freed when it  
>> shouldn't be) which causes a segfault on the second or third  
>> session after the module being loaded.
>>
>> Without wishing to sound like a stuck record, one thing that you  
>> guys really ought to do is to decide what's supported and what  
>> isn't, and make this obvious - for example, move unsupported  
>> modules to a different place in the tree, don't have them built by  
>> default, etc.  MRCP is in the specsheet on the Wiki.  Otherwise  
>> folk like Mark and I spend time installing stuff, go round in  
>> circles a bit trying to make it work, and then find out (a) that it  
>> doesn't and (b) it's not going to be fixed because it's not  
>> supported.
>>
>> Cheers --
>>
>> Dave
>>> I would not say it is totally broken, it is known to work in quite  
>>> a few places, but we are unlikely to be doing any new fixes in it.
>>>
>>> Mike
>>>
>>> On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>>
>>>> Hi Anthony,
>>>>
>>>> Oh! OK.
>>>>
>>>> So is this module "totally broken".
>>>>
>>>> I say this because I can't seem to get it to work at all with the  
>>>> example in that Mod_openmrcp wiki page but I thought it might  
>>>> because I'm not be using the right Cepstral software (freetrial  
>>>> download versus the paided for SDK) or that I'm not using the  
>>>> right port numbers or something else I didn't do. I used TcpView  
>>>> to look at local port associated with my Cepstral software and  
>>>> changed a few things but still nothing. I changed the loglevel  
>>>> setting to 7 in the wiki's example but I don't see the kind of  
>>>> output on the console that I would expect for debug mode.
>>>>
>>>> Thanks. Mark.
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> -- 
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>> http://www.3c.co.uk
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> -- 
> David Knell, Director, 3C Limited
> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
> http://www.3c.co.uk
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Dec  2 08:32:31 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 10:32:31 -0600
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
Message-ID: <191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>

Naturally, either way is stupid.

The whole idea of putting the transport in a uri param is equally stupid to
using 2 different protocol names but since SIP is the descendant of http it
they decided to stick with the stupidity of http/https and have sip/sips
which is almost as if it was designed to break all software trying to keep
up with url syntax.

If they are going to insist on using text params you'd think something like
transport=foo;security=tls would be even *more* flexable in case alternate
methods to encrypt crop up.

This is, of course, the first step into a lengthy 12 hour discussion on how
stupid SIP and url/text based
protocols are.

I dare someone to crank up the pcap on a box doing SIP presence for 20
phones and "read"
the 1200 byte messages with all kinds of hyeroglyphic url syntax and
embedded xml payloads and write
up a paper on how much "sense" it makes to have it be "readable".

PS

supposedly sofia can support sctp,
someone should try it.



On Mon, Dec 1, 2008 at 9:43 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On 12/1/08, Thomas Troy <ttroy50 at gmail.com> wrote:
> ..snip..
> >
> > Out of interest do you have any links to anywhere this is discussed in
> terms
> > of general sip implementations?
> >
>
> Uh oh, here we go again...
>
> http://www.iana.org/assignments/sip-parameters
> http://tools.ietf.org/html/rfc3969
>
>
> https://lists.cs.columbia.edu/pipermail/sip-implementors/2005-August/010047.html
>
> Implementation wise, most devices tend to use transport=tls:
>
> SIPFoundry - From what I've seen
> Snom
> SERs
> Asterisk (If you are using TLS)
> Cisco - I *believe* you can use either a SIPS URI or the transport=tls
> parameter for various SIP targets
>
>  As the RFC (basically) states (RFC3261, section 12.1.x),
> transport=tls was deprecated in RFC 3261 because you should also be
> able to do TLS over SCTP (RFC3436), which makes transport=tls a bit
> ambiguous. sips:user at domain;transport=tcp or
> sips:user at domain;transport=sctp is a bit more flexible.
>
>  I don't know if I've ever seen anything default to SIPS URIs.  I
> also don't think I've ever specifically tried using them.  However, my
> experience with TLS is admittedly somewhat limited so this shouldn't
> be taken as gospel.  As you can see from the discussions on
> sip-implementors, this gets interesting when different devices are
> traversing a proxy using different URI schemes...
>
>  However, I suspect this won't become an issue until most SIP
> implementations support SCTP.  That should be exciting! ;)
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Tue Dec  2 08:43:44 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 10:43:44 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
	<191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
	<39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>
Message-ID: <191c3a030812020843m1bed6ab2mf77f1132ec1f26fa@mail.gmail.com>

from build root:

svn co -r8809
http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvoxsrc/mod/asr_tts/mod_lumenvox


They did seem to express an interest in granting some dev licenses when they
realized we took the code out of tree but I have not actually dealt with the
issue yet because I have been overwhelmed.

I don't know if this code works anymore with the latest revision of the api
but there it is.





On Tue, Dec 2, 2008 at 10:24 AM, Andrew Gilbert <gilbertandrew at me.com>wrote:

> Mark and David,
>
> I am willing to help some with testing here as well, if you need it. Ping
> me directly or we can get on the IRC. I am on Mac OS, but have readily
> available vm's with Debian, etc. I also have Prophecy.
>
> I have a general interest in an ASR solution as well. Voxeo is great, but
> using it as an MRCP proxy seems odd. As a full fledged VXML solution it is
> great, if you can afford it. But having a good ASR solution is good first
> step to trying to get something like OpenVXI working as well.
>
> That said, seems like a bounty or money to help FS is a better spend
> anyway. It is a one time cost, not a variable cost. And it goes straight to
> the guys doing the real work.
>
> I built unimrcp last night, it was quite straight forward. In theory, if I
> weren't old and my C/autoconf skills rather atrophied, it wouldn't seem like
> it would be that huge a deal to port/fix openmrcp to unimrcp.
>
> Finally, Anthony I was looking at the Lumenvox path as well, but got
> deterred by the licensing hassle. This seems to be a universal ASR issue. I
> would reason I can find the old module in SVN? Were they going to grant
> "community dev" licenses? Again - I am willing to volunteer to do some
> testing/doc at least.
>
> Andy
>
>
>
> On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote:
>
> If you can get it to break on linux I will ssh in and fix it for you.
> If you cannot, i can try to fix it for you over rdp but that won't be very
> fun.
>
> We can think about reinstating mod_lumenvox as well as another windows
> based asr
> alternative.  I deleted it for the same reason we will probably delete
> mod_openmrcp because
> nobody was using it and there was no way to support it because our dev
> licenses had expired.
>
> Lumenvox has offered us some new dev licenses to bring it back but I would
> need someone to actually want it to work to put in charge of it.
>
> We will be clear about what is supported and what is not in the 1.0.2
> release scheduled
> to be released in the near future.
>
>
>
>
> On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:
>
>>  Hi Anthony,
>>
>> mod_openmrcp was a contribution to the community by a 3rd party
>> individual.
>>
>> As i have clearly stated in 2 previous emails, the man has decided to
>> discontinue the openmrcp project.
>> So now we are left with the remains of the module and discontinued code.
>> This was not our decision it was his.
>>
>> I absolutely understand this but it's important, from a user point of
>> view, to be able to know which bits of FS are current/supported and which
>> aren't.
>>
>> Some people use it without issue which may mean that the crash you
>> reported is windows specific and I do not have a working lab of any mrcp
>> capbable system to try it against in unix for that matter.  I have a list of
>> work to do from here to the moon and back so on an issue like this, unless
>> someone can hand me login credentials to some box and give me a phone number
>> to dial to reporduce the issue, it will be a long time until we can deal
>> with it.
>>
>> It's useful to know that there are people using mod_openmrcp without
>> issue: I did ask here if anyone was a while back, and no-one fessed up.
>> I'll give it a go on a Linux box and report back.  And if you'd like a
>> dev/test environment set up, then just tell me which one.
>>
>> And the question arises, should we bother working on it anymore if the lib
>> has been abandoned and we cannot even get any support from it's author which
>> is where the problem most likely lies.
>>
>> I try not to get too annoyed by these remarks about what we *ought to do*
>> because I know people lose sight of how much of the work to support the
>> project is done by a small group of 3 people and not the 2000 people it
>> appears to be from the outside looking in. (I've been answering email for 4
>> hours now)
>>
>> Those guys who claim to have all that money in an offshore bank account
>> are lying - you don't have to reply to them in future ;-)  Seriously,
>> though, I don't think it's too outrageous an idea to document what's
>> supported and were you (for example) to have suggested that I get in touch
>> with the contributors to the various modules, ask them what their view of
>> its status is, condense the answers in to a list and report back, it's
>> something I'd quite happily do.
>>
>> My suggestion is to pool some cash and pay the guy to make mod_unimrcp for
>> FS that we can maintain in tree knowing the development can be supported by
>> the original author.
>>
>> Quite happy to participate in that, too.. the problem is that I've a demo
>> to do like yesterday and the timescale for mod_unimrcp is a bit on the long
>> side for that.  I'd rather not have to do it with Asterisk and Lumenvox..!
>>
>> Cheers --
>>
>> Dave
>>
>>
>>
>> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>>
>>>  Hi Mike,
>>>
>>> My experience is that it's somewhat broken - it took two trivial tweaks
>>> to get it to work with IBM's ASR and TTS, but there's a more intractable
>>> problem to do with memory getting overwritten (I assume that this is
>>> something to do with something being freed when it shouldn't be) which
>>> causes a segfault on the second or third session after the module being
>>> loaded.
>>>
>>> Without wishing to sound like a stuck record, one thing that you guys
>>> really ought to do is to decide what's supported and what isn't, and make
>>> this obvious - for example, move unsupported modules to a different place in
>>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>>> round in circles a bit trying to make it work, and then find out (a) that it
>>> doesn't and (b) it's not going to be fixed because it's not supported.
>>>
>>> Cheers --
>>>
>>> Dave
>>>
>>>  I would not say it is totally broken, it is known to work in quite a
>>> few places, but we are unlikely to be doing any new fixes in it.
>>>  Mike
>>>
>>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>>
>>>  Hi Anthony,
>>>
>>> Oh! OK.
>>>
>>> So is this module "totally broken".
>>>
>>> I say this because I can't seem to get it to work at all with the example
>>> in that Mod_openmrcp wiki page but I thought it might because I'm not be
>>> using the right Cepstral software (freetrial download versus the paided for
>>> SDK) or that I'm not using the right port numbers or something else I didn't
>>> do. I used TcpView to look at local port associated with my Cepstral
>>> software and changed a few things but still nothing. I changed the loglevel
>>> setting to 7 in the wiki's example but I don't see the kind of output on the
>>> console that I would expect for debug mode.
>>>
>>> Thanks. Mark.
>>>
>>>
>>>    ------------------------------
>>> _______________________________________________
>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> David Knell, Director, 3C Limited
>>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> ------------------------------
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gilbertandrew at me.com  Tue Dec  2 09:27:25 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 02 Dec 2008 12:27:25 -0500
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <191c3a030812020843m1bed6ab2mf77f1132ec1f26fa@mail.gmail.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<191c3a030812010937k380ca578h2b5ef6f8766c3588@mail.gmail.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
	<191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
	<39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>
	<191c3a030812020843m1bed6ab2mf77f1132ec1f26fa@mail.gmail.com>
Message-ID: <7ED7E135-711F-4E7F-BB3B-6B6014211B90@me.com>

Ok

I have a ping in with Lumenvox about dev licensing, and pulled the  
mod. Not sure where this will go, but will take a peek at things.  
Balancing the effort against something like getting unimcrp going and/ 
or openmrcp tested and stable.

Thanks.

Andy


On Dec 2, 2008, at 11:43 AM, Anthony Minessale wrote:

> from build root:
>
> svn co -r8809 http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvox 
>  src/mod/asr_tts/mod_lumenvox
>
>
> They did seem to express an interest in granting some dev licenses  
> when they realized we took the code out of tree but I have not  
> actually dealt with the issue yet because I have been overwhelmed.
>
> I don't know if this code works anymore with the latest revision of  
> the api but there it is.
>
>
>
>
>
> On Tue, Dec 2, 2008 at 10:24 AM, Andrew Gilbert  
> <gilbertandrew at me.com> wrote:
> Mark and David,
>
> I am willing to help some with testing here as well, if you need it.  
> Ping me directly or we can get on the IRC. I am on Mac OS, but have  
> readily available vm's with Debian, etc. I also have Prophecy.
>
> I have a general interest in an ASR solution as well. Voxeo is  
> great, but using it as an MRCP proxy seems odd. As a full fledged  
> VXML solution it is great, if you can afford it. But having a good  
> ASR solution is good first step to trying to get something like  
> OpenVXI working as well.
>
> That said, seems like a bounty or money to help FS is a better spend  
> anyway. It is a one time cost, not a variable cost. And it goes  
> straight to the guys doing the real work.
>
> I built unimrcp last night, it was quite straight forward. In  
> theory, if I weren't old and my C/autoconf skills rather atrophied,  
> it wouldn't seem like it would be that huge a deal to port/fix  
> openmrcp to unimrcp.
>
> Finally, Anthony I was looking at the Lumenvox path as well, but got  
> deterred by the licensing hassle. This seems to be a universal ASR  
> issue. I would reason I can find the old module in SVN? Were they  
> going to grant "community dev" licenses? Again - I am willing to  
> volunteer to do some testing/doc at least.
>
> Andy
>
>
>
> On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote:
>
>> If you can get it to break on linux I will ssh in and fix it for you.
>> If you cannot, i can try to fix it for you over rdp but that won't  
>> be very fun.
>>
>> We can think about reinstating mod_lumenvox as well as another  
>> windows based asr
>> alternative.  I deleted it for the same reason we will probably  
>> delete mod_openmrcp because
>> nobody was using it and there was no way to support it because our  
>> dev licenses had expired.
>>
>> Lumenvox has offered us some new dev licenses to bring it back but  
>> I would need someone to actually want it to work to put in charge  
>> of it.
>>
>> We will be clear about what is supported and what is not in the  
>> 1.0.2 release scheduled
>> to be released in the near future.
>>
>>
>>
>>
>> On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:
>> Hi Anthony,
>>
>>> mod_openmrcp was a contribution to the community by a 3rd party  
>>> individual.
>>>
>>> As i have clearly stated in 2 previous emails, the man has decided  
>>> to discontinue the openmrcp project.
>>> So now we are left with the remains of the module and discontinued  
>>> code.  This was not our decision it was his.
>> I absolutely understand this but it's important, from a user point  
>> of view, to be able to know which bits of FS are current/supported  
>> and which aren't.
>>
>>> Some people use it without issue which may mean that the crash you  
>>> reported is windows specific and I do not have a working lab of  
>>> any mrcp capbable system to try it against in unix for that  
>>> matter.  I have a list of work to do from here to the moon and  
>>> back so on an issue like this, unless someone can hand me login  
>>> credentials to some box and give me a phone number to dial to  
>>> reporduce the issue, it will be a long time until we can deal with  
>>> it.
>> It's useful to know that there are people using mod_openmrcp  
>> without issue: I did ask here if anyone was a while back, and no- 
>> one fessed up.  I'll give it a go on a Linux box and report back.   
>> And if you'd like a dev/test environment set up, then just tell me  
>> which one.
>>
>>> And the question arises, should we bother working on it anymore if  
>>> the lib has been abandoned and we cannot even get any support from  
>>> it's author which is where the problem most likely lies.
>>>
>>> I try not to get too annoyed by these remarks about what we *ought  
>>> to do* because I know people lose sight of how much of the work to  
>>> support the project is done by a small group of 3 people and not  
>>> the 2000 people it appears to be from the outside looking in.  
>>> (I've been answering email for 4 hours now)
>> Those guys who claim to have all that money in an offshore bank  
>> account are lying - you don't have to reply to them in future ;-)   
>> Seriously, though, I don't think it's too outrageous an idea to  
>> document what's supported and were you (for example) to have  
>> suggested that I get in touch with the contributors to the various  
>> modules, ask them what their view of its status is, condense the  
>> answers in to a list and report back, it's something I'd quite  
>> happily do.
>>
>>> My suggestion is to pool some cash and pay the guy to make  
>>> mod_unimrcp for FS that we can maintain in tree knowing the  
>>> development can be supported by the original author.
>> Quite happy to participate in that, too.. the problem is that I've  
>> a demo to do like yesterday and the timescale for mod_unimrcp is a  
>> bit on the long side for that.  I'd rather not have to do it with  
>> Asterisk and Lumenvox..!
>>
>> Cheers --
>>
>> Dave
>>
>>>
>>>
>>> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>>> Hi Mike,
>>>
>>> My experience is that it's somewhat broken - it took two trivial  
>>> tweaks to get it to work with IBM's ASR and TTS, but there's a  
>>> more intractable problem to do with memory getting overwritten (I  
>>> assume that this is something to do with something being freed  
>>> when it shouldn't be) which causes a segfault on the second or  
>>> third session after the module being loaded.
>>>
>>> Without wishing to sound like a stuck record, one thing that you  
>>> guys really ought to do is to decide what's supported and what  
>>> isn't, and make this obvious - for example, move unsupported  
>>> modules to a different place in the tree, don't have them built by  
>>> default, etc.  MRCP is in the specsheet on the Wiki.  Otherwise  
>>> folk like Mark and I spend time installing stuff, go round in  
>>> circles a bit trying to make it work, and then find out (a) that  
>>> it doesn't and (b) it's not going to be fixed because it's not  
>>> supported.
>>>
>>> Cheers --
>>>
>>> Dave
>>>> I would not say it is totally broken, it is known to work in  
>>>> quite a few places, but we are unlikely to be doing any new fixes  
>>>> in it.
>>>>
>>>> Mike
>>>>
>>>> On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>>>
>>>>> Hi Anthony,
>>>>>
>>>>> Oh! OK.
>>>>>
>>>>> So is this module "totally broken".
>>>>>
>>>>> I say this because I can't seem to get it to work at all with  
>>>>> the example in that Mod_openmrcp wiki page but I thought it  
>>>>> might because I'm not be using the right Cepstral software  
>>>>> (freetrial download versus the paided for SDK) or that I'm not  
>>>>> using the right port numbers or something else I didn't do. I  
>>>>> used TcpView to look at local port associated with my Cepstral  
>>>>> software and changed a few things but still nothing. I changed  
>>>>> the loglevel setting to 7 in the wiki's example but I don't see  
>>>>> the kind of output on the console that I would expect for debug  
>>>>> mode.
>>>>>
>>>>> Thanks. Mark.
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> -- 
>>> David Knell, Director, 3C Limited
>>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>>> http://www.3c.co.uk
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> -- 
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:213-799-1400
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> -- 
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>> http://www.3c.co.uk
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From odermann at googlemail.com  Tue Dec  2 09:40:45 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 2 Dec 2008 18:40:45 +0100
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
Message-ID: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>

hi,

because we do not get tired of testing and playing a lot with the
beloved fs, we now arrived at the fax feature :-)

i am not sure if the docs are up to date or if there was a lot of
development in the meantime. therefore i would like to ask, what is
possible and what will come in the near future.

we are using fs, socket outbound and php and would like to make
something like fax to mail as an additional service.

is t38 supported?
can i pass incoming faxes over the same socket as calls?
can i convert faxes into pdf?
is fax over sip reliable (as far as i have heard, under asterisk fax
is nothing one should use)?
and so on, and so on....

i would be very happy to hear some user experiences with fs and fax.
if it seems, that we can use fax with over socket outbound, we will do
hardcore testing ;-)

thanks,
dennis



From anthony.minessale at gmail.com  Tue Dec  2 09:58:59 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 11:58:59 -0600
Subject: [Freeswitch-users] Problems with Mod_openMRCP
In-Reply-To: <7ED7E135-711F-4E7F-BB3B-6B6014211B90@me.com>
References: <8CB1F9C694E75D9-180-518@mblk-d51.sysops.aol.com>
	<8CB21F883CFBBE5-CA4-A88@MBLK-M05.sysops.aol.com>
	<CBEEFAF8-AF75-4958-AE56-D591AB3CCA2A@jerris.com>
	<4934323B.9000305@3c.co.uk>
	<191c3a030812011117p205685a4k12937a301d8b1374@mail.gmail.com>
	<4934D197.8080007@3c.co.uk>
	<191c3a030812020751y6f2b69b6neb4cc0197be745f6@mail.gmail.com>
	<39BD3113-F17F-4AD5-AEBD-776DC4F3EE8A@me.com>
	<191c3a030812020843m1bed6ab2mf77f1132ec1f26fa@mail.gmail.com>
	<7ED7E135-711F-4E7F-BB3B-6B6014211B90@me.com>
Message-ID: <191c3a030812020958s30d0e6d2ub3452e7e63fa19d9@mail.gmail.com>

They contacted us shortly thereafter and asked if we want to have them sell
you the license for 50 bucks.
hmm, i wonder why i deleted the module.....

I will tell them that if they give you a developer license you will work on
getting it back into trunk.

On Tue, Dec 2, 2008 at 11:27 AM, Andrew Gilbert <gilbertandrew at me.com>wrote:

> Ok
>
> I have a ping in with Lumenvox about dev licensing, and pulled the mod. Not
> sure where this will go, but will take a peek at things. Balancing the
> effort against something like getting unimcrp going and/or openmrcp tested
> and stable.
>
> Thanks.
>
> Andy
>
>
> On Dec 2, 2008, at 11:43 AM, Anthony Minessale wrote:
>
> from build root:
>
> svn co -r8809
> http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/asr_tts/mod_lumenvoxsrc/mod/asr_tts/mod_lumenvox
>
>
> They did seem to express an interest in granting some dev licenses when
> they realized we took the code out of tree but I have not actually dealt
> with the issue yet because I have been overwhelmed.
>
> I don't know if this code works anymore with the latest revision of the api
> but there it is.
>
>
>
>
>
> On Tue, Dec 2, 2008 at 10:24 AM, Andrew Gilbert <gilbertandrew at me.com>wrote:
>
>> Mark and David,
>>
>> I am willing to help some with testing here as well, if you need it. Ping
>> me directly or we can get on the IRC. I am on Mac OS, but have readily
>> available vm's with Debian, etc. I also have Prophecy.
>>
>> I have a general interest in an ASR solution as well. Voxeo is great, but
>> using it as an MRCP proxy seems odd. As a full fledged VXML solution it is
>> great, if you can afford it. But having a good ASR solution is good first
>> step to trying to get something like OpenVXI working as well.
>>
>> That said, seems like a bounty or money to help FS is a better spend
>> anyway. It is a one time cost, not a variable cost. And it goes straight to
>> the guys doing the real work.
>>
>> I built unimrcp last night, it was quite straight forward. In theory, if I
>> weren't old and my C/autoconf skills rather atrophied, it wouldn't seem like
>> it would be that huge a deal to port/fix openmrcp to unimrcp.
>>
>> Finally, Anthony I was looking at the Lumenvox path as well, but got
>> deterred by the licensing hassle. This seems to be a universal ASR issue. I
>> would reason I can find the old module in SVN? Were they going to grant
>> "community dev" licenses? Again - I am willing to volunteer to do some
>> testing/doc at least.
>>
>> Andy
>>
>>
>>
>> On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote:
>>
>> If you can get it to break on linux I will ssh in and fix it for you.
>> If you cannot, i can try to fix it for you over rdp but that won't be very
>> fun.
>>
>> We can think about reinstating mod_lumenvox as well as another windows
>> based asr
>> alternative.  I deleted it for the same reason we will probably delete
>> mod_openmrcp because
>> nobody was using it and there was no way to support it because our dev
>> licenses had expired.
>>
>> Lumenvox has offered us some new dev licenses to bring it back but I would
>> need someone to actually want it to work to put in charge of it.
>>
>> We will be clear about what is supported and what is not in the 1.0.2
>> release scheduled
>> to be released in the near future.
>>
>>
>>
>>
>> On Tue, Dec 2, 2008 at 12:11 AM, David Knell <dave at 3c.co.uk> wrote:
>>
>>>  Hi Anthony,
>>>
>>> mod_openmrcp was a contribution to the community by a 3rd party
>>> individual.
>>>
>>> As i have clearly stated in 2 previous emails, the man has decided to
>>> discontinue the openmrcp project.
>>> So now we are left with the remains of the module and discontinued code.
>>> This was not our decision it was his.
>>>
>>>  I absolutely understand this but it's important, from a user point of
>>> view, to be able to know which bits of FS are current/supported and which
>>> aren't.
>>>
>>> Some people use it without issue which may mean that the crash you
>>> reported is windows specific and I do not have a working lab of any mrcp
>>> capbable system to try it against in unix for that matter.  I have a list of
>>> work to do from here to the moon and back so on an issue like this, unless
>>> someone can hand me login credentials to some box and give me a phone number
>>> to dial to reporduce the issue, it will be a long time until we can deal
>>> with it.
>>>
>>>  It's useful to know that there are people using mod_openmrcp without
>>> issue: I did ask here if anyone was a while back, and no-one fessed up.
>>> I'll give it a go on a Linux box and report back.  And if you'd like a
>>> dev/test environment set up, then just tell me which one.
>>>
>>> And the question arises, should we bother working on it anymore if the
>>> lib has been abandoned and we cannot even get any support from it's author
>>> which is where the problem most likely lies.
>>>
>>> I try not to get too annoyed by these remarks about what we *ought to do*
>>> because I know people lose sight of how much of the work to support the
>>> project is done by a small group of 3 people and not the 2000 people it
>>> appears to be from the outside looking in. (I've been answering email for 4
>>> hours now)
>>>
>>>  Those guys who claim to have all that money in an offshore bank account
>>> are lying - you don't have to reply to them in future ;-)  Seriously,
>>> though, I don't think it's too outrageous an idea to document what's
>>> supported and were you (for example) to have suggested that I get in touch
>>> with the contributors to the various modules, ask them what their view of
>>> its status is, condense the answers in to a list and report back, it's
>>> something I'd quite happily do.
>>>
>>> My suggestion is to pool some cash and pay the guy to make mod_unimrcp
>>> for FS that we can maintain in tree knowing the development can be supported
>>> by the original author.
>>>
>>> Quite happy to participate in that, too.. the problem is that I've a demo
>>> to do like yesterday and the timescale for mod_unimrcp is a bit on the long
>>> side for that.  I'd rather not have to do it with Asterisk and Lumenvox..!
>>>
>>> Cheers --
>>>
>>> Dave
>>>
>>>
>>>
>>> On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave at 3c.co.uk> wrote:
>>>
>>>>  Hi Mike,
>>>>
>>>> My experience is that it's somewhat broken - it took two trivial tweaks
>>>> to get it to work with IBM's ASR and TTS, but there's a more intractable
>>>> problem to do with memory getting overwritten (I assume that this is
>>>> something to do with something being freed when it shouldn't be) which
>>>> causes a segfault on the second or third session after the module being
>>>> loaded.
>>>>
>>>> Without wishing to sound like a stuck record, one thing that you guys
>>>> really ought to do is to decide what's supported and what isn't, and make
>>>> this obvious - for example, move unsupported modules to a different place in
>>>> the tree, don't have them built by default, etc.  MRCP is in the specsheet
>>>> on the Wiki.  Otherwise folk like Mark and I spend time installing stuff, go
>>>> round in circles a bit trying to make it work, and then find out (a) that it
>>>> doesn't and (b) it's not going to be fixed because it's not supported.
>>>>
>>>> Cheers --
>>>>
>>>> Dave
>>>>
>>>>  I would not say it is totally broken, it is known to work in quite a
>>>> few places, but we are unlikely to be doing any new fixes in it.
>>>>  Mike
>>>>
>>>>  On Dec 1, 2008, at 1:19 PM, mszlazak at aol.com wrote:
>>>>
>>>>  Hi Anthony,
>>>>
>>>> Oh! OK.
>>>>
>>>> So is this module "totally broken".
>>>>
>>>> I say this because I can't seem to get it to work at all with the
>>>> example in that Mod_openmrcp wiki page but I thought it might because I'm
>>>> not be using the right Cepstral software (freetrial download versus the
>>>> paided for SDK) or that I'm not using the right port numbers or something
>>>> else I didn't do. I used TcpView to look at local port associated with my
>>>> Cepstral software and changed a few things but still nothing. I changed the
>>>> loglevel setting to 7 in the wiki's example but I don't see the kind of
>>>> output on the console that I would expect for debug mode.
>>>>
>>>> Thanks. Mark.
>>>>
>>>>
>>>>    ------------------------------
>>>> _______________________________________________
>>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> --
>>>> David Knell, Director, 3C Limited
>>>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> ------------------------------
>>> _______________________________________________
>>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> David Knell, Director, 3C Limited
>>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031http://www.3c.co.uk
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From sergey.kirillov at gmail.com  Tue Dec  2 10:06:15 2008
From: sergey.kirillov at gmail.com (Sergey Kirillov)
Date: Tue, 02 Dec 2008 20:06:15 +0200
Subject: [Freeswitch-users] Support for Junghanns duoBRI
Message-ID: <49357917.30804@gmail.com>

Cool. Thanks for the answer.

> All HFC-based cards supported by bristuffed Zaptel should work.
>
> Stefan



From kkielhofner at star2star.com  Tue Dec  2 11:03:41 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Tue, 2 Dec 2008 14:03:41 -0500
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
	<191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
Message-ID: <2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>

On 12/2/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Naturally, either way is stupid.

  Word.

> The whole idea of putting the transport in a uri param is equally stupid to
> using 2 different protocol names but since SIP is the descendant of http it
> they decided to stick with the stupidity of http/https and have sip/sips
> which is almost as if it was designed to break all software trying to keep
> up with url syntax.

  Too late now.

> If they are going to insist on using text params you'd think something like
> transport=foo;security=tls would be even *more* flexable in case alternate
> methods to encrypt crop up.

  I can agree with you here...

  URI parameters in SIP have come to be the catch all for random junk
that doesn't seem to fit anywhere else.  Note that "random junk"
includes everything from transport, to number portability, to CICs, to
ISUP-OLI and on.

  Even in my world setting up proxies, UAs, etc to parse out the
various crap people put in SIP URI params is a hassle.  A big one.

  What a mess!!!

> This is, of course, the first step into a lengthy 12 hour discussion on how
> stupid SIP and url/text based
>  protocols are.

  I like them but I'm weird.

> I dare someone to crank up the pcap on a box doing SIP presence for 20
> phones and "read"
> the 1200 byte messages with all kinds of hyeroglyphic url syntax and
> embedded xml payloads and write
>  up a paper on how much "sense" it makes to have it be "readable".

  I do it all the time.  I think it's quite usable.  ngrep provides a
small enough binary and the ability to match on text.  Certainly
easier to use, especially on embedded systems without the luxury of
dedicated protocol decoders.  With a simple ngrep binary I can debug
any text based protocol I understand.

  Of course, turn on TLS and see how useful *any* of these tools are...

  The core SIP spec and authors can't be blamed for the various junk
people have been putting in SIP bodies.  If what's going on in the
real world is any indication, that ship sailed long ago.  At this
point as long as implementations can at least handle multi-part
sensibly and everyone specifies the correct MIME type I don't really
care.

  Even nastier examples abound - embedded, encapsulated ISUP!  How
about GTD?  What about Linksys phones using SIP INFO to serve
directories?  Man I could go on and on...

  I'm not going to write a paper about it but I don't think it's that
bad.  Maybe I'm not just weird; maybe I'm a masochist! :)

> PS
>
> supposedly sofia can support sctp,
> someone should try it.

  That would be cool.  For anyone wanting to try, various SERs support
SCTP.  Cisco gateways do too.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From c_cav_01 at yahoo.com  Tue Dec  2 11:08:13 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 11:08:13 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>
References: <20791453.post@talk.nabble.com>
	<B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>
Message-ID: <20798791.post@talk.nabble.com>


-- filename "dialplan/extensions/13033253678.xml" --   This is the primary
DID assigned.

  <extension name="3033253678">
    <!--<condition field="destination_number"
expression="^[0,1]?3033253678$"/>-->
    <condition field="${sip_to_user}" expression="^([0,1]?)(3033253678)$">
      <action application="ring_ready"/>
      <action application="set" data="${dialed_user}=$2" />
      <action application="set" data="ignore_early_media=true"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESINATION,NO_ANSWER,DESTINATION_OUT_OF_ORDER,USER_BUSY"/>
      <action application="playback"
data="/usr/local/freeswitch/sounds/holdmusic/06.mp3"/>
      <action application="set" data="call_timeout=25"/>
      <action application="bridge" data="user/$2@$${domain}"/>
      <action application="answer"/>
      <action application="voicemail" data="default $${domain} $2"/>
    </condition>
  </extension>
  

-- filename "dialplan/extensions/18162565804.xml" --   This is the primary
DID assigned.

<extension name="8162565804">
    <!--<condition field="destination_number"
expression="^[0,1]?3033253678$"/>-->
    <condition field="${sip_to_user}" expression="^([0,1]?)(8162565804)$">
      <action application="ring_ready"/>
      <action application="set" data="ignore_early_media=true"/>
      <action application="set" data="call_timeout=25"/>
      <action application="set" data="${dialed_user}=$2"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESINATION,NO_ANSWER,DESTINATION_OUT_OF_ORDER"/>
      <action application="playback"
data="/usr/local/freeswitch/sounds/holdmusic/07.mp3"/>
      <action application="bridge" data="user/$2@$${domain}"/>
      <action application="answer"/>
      <action application="voicemail" data="default $${domain} $2"/>
    </condition>
  </extension>

-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20798791.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From msc at freeswitch.org  Tue Dec  2 11:13:46 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 11:13:46 -0800
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
	<191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
	<2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
Message-ID: <87f2f3b90812021113x7a7c5c51v11e80a6ef82c012d@mail.gmail.com>

Bring on SNAP, baby!

On Tue, Dec 2, 2008 at 11:03 AM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On 12/2/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> > Naturally, either way is stupid.
>
>  Word.
>
> > The whole idea of putting the transport in a uri param is equally stupid
> to
> > using 2 different protocol names but since SIP is the descendant of http
> it
> > they decided to stick with the stupidity of http/https and have sip/sips
> > which is almost as if it was designed to break all software trying to
> keep
> > up with url syntax.
>
>  Too late now.
>
> > If they are going to insist on using text params you'd think something
> like
> > transport=foo;security=tls would be even *more* flexable in case
> alternate
> > methods to encrypt crop up.
>
>  I can agree with you here...
>
>  URI parameters in SIP have come to be the catch all for random junk
> that doesn't seem to fit anywhere else.  Note that "random junk"
> includes everything from transport, to number portability, to CICs, to
> ISUP-OLI and on.
>
>  Even in my world setting up proxies, UAs, etc to parse out the
> various crap people put in SIP URI params is a hassle.  A big one.
>
>  What a mess!!!
>
> > This is, of course, the first step into a lengthy 12 hour discussion on
> how
> > stupid SIP and url/text based
> >  protocols are.
>
>  I like them but I'm weird.
>
> > I dare someone to crank up the pcap on a box doing SIP presence for 20
> > phones and "read"
> > the 1200 byte messages with all kinds of hyeroglyphic url syntax and
> > embedded xml payloads and write
> >  up a paper on how much "sense" it makes to have it be "readable".
>
>  I do it all the time.  I think it's quite usable.  ngrep provides a
> small enough binary and the ability to match on text.  Certainly
> easier to use, especially on embedded systems without the luxury of
> dedicated protocol decoders.  With a simple ngrep binary I can debug
> any text based protocol I understand.
>
>  Of course, turn on TLS and see how useful *any* of these tools are...
>
>  The core SIP spec and authors can't be blamed for the various junk
> people have been putting in SIP bodies.  If what's going on in the
> real world is any indication, that ship sailed long ago.  At this
> point as long as implementations can at least handle multi-part
> sensibly and everyone specifies the correct MIME type I don't really
> care.
>
>  Even nastier examples abound - embedded, encapsulated ISUP!  How
> about GTD?  What about Linksys phones using SIP INFO to serve
> directories?  Man I could go on and on...
>
>  I'm not going to write a paper about it but I don't think it's that
> bad.  Maybe I'm not just weird; maybe I'm a masochist! :)
>
> > PS
> >
> > supposedly sofia can support sctp,
> > someone should try it.
>
>  That would be cool.  For anyone wanting to try, various SERs support
> SCTP.  Cisco gateways do too.
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Dec  2 11:16:36 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 13:16:36 -0600
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
	<191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
	<2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
Message-ID: <191c3a030812021116n724f77c8oe477a1585f12e8da@mail.gmail.com>

We'll schedule a round table with the topic

SIP OMFG STFU

At the next ClueCon aug 4th-6th 2009 to stir things up a bit =D


On Tue, Dec 2, 2008 at 1:03 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On 12/2/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> > Naturally, either way is stupid.
>
>   Word.
>
> > The whole idea of putting the transport in a uri param is equally stupid
> to
> > using 2 different protocol names but since SIP is the descendant of http
> it
> > they decided to stick with the stupidity of http/https and have sip/sips
> > which is almost as if it was designed to break all software trying to
> keep
> > up with url syntax.
>
>   Too late now.
>
> > If they are going to insist on using text params you'd think something
> like
> > transport=foo;security=tls would be even *more* flexable in case
> alternate
> > methods to encrypt crop up.
>
>   I can agree with you here...
>
>  URI parameters in SIP have come to be the catch all for random junk
> that doesn't seem to fit anywhere else.  Note that "random junk"
> includes everything from transport, to number portability, to CICs, to
> ISUP-OLI and on.
>
>  Even in my world setting up proxies, UAs, etc to parse out the
> various crap people put in SIP URI params is a hassle.  A big one.
>
>  What a mess!!!
>
> > This is, of course, the first step into a lengthy 12 hour discussion on
> how
> > stupid SIP and url/text based
> >  protocols are.
>
>   I like them but I'm weird.
>
> > I dare someone to crank up the pcap on a box doing SIP presence for 20
> > phones and "read"
> > the 1200 byte messages with all kinds of hyeroglyphic url syntax and
> > embedded xml payloads and write
> >  up a paper on how much "sense" it makes to have it be "readable".
>
>   I do it all the time.  I think it's quite usable.  ngrep provides a
> small enough binary and the ability to match on text.  Certainly
> easier to use, especially on embedded systems without the luxury of
> dedicated protocol decoders.  With a simple ngrep binary I can debug
> any text based protocol I understand.
>
>  Of course, turn on TLS and see how useful *any* of these tools are...
>
>  The core SIP spec and authors can't be blamed for the various junk
> people have been putting in SIP bodies.  If what's going on in the
> real world is any indication, that ship sailed long ago.  At this
> point as long as implementations can at least handle multi-part
> sensibly and everyone specifies the correct MIME type I don't really
> care.
>
>  Even nastier examples abound - embedded, encapsulated ISUP!  How
> about GTD?  What about Linksys phones using SIP INFO to serve
> directories?  Man I could go on and on...
>
>  I'm not going to write a paper about it but I don't think it's that
> bad.  Maybe I'm not just weird; maybe I'm a masochist! :)
>
> > PS
> >
> > supposedly sofia can support sctp,
> > someone should try it.
>
>   That would be cool.  For anyone wanting to try, various SERs support
> SCTP.  Cisco gateways do too.
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kkielhofner at star2star.com  Tue Dec  2 11:20:31 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Tue, 2 Dec 2008 14:20:31 -0500
Subject: [Freeswitch-users] TLS receiving calls
In-Reply-To: <191c3a030812021116n724f77c8oe477a1585f12e8da@mail.gmail.com>
References: <20771637.post@talk.nabble.com>
	<EDE60C32-130E-4243-B413-86EF58F0B81F@freeswitch.org>
	<adbe60280812010849w6e5741dei5cf14842003bdb0e@mail.gmail.com>
	<2d9149cd0812011943k2f79b6f2if61ba33d8e66a394@mail.gmail.com>
	<191c3a030812020832x628e627cm21337495dcfc48f7@mail.gmail.com>
	<2d9149cd0812021103r4b7f0988s4c571f158f19c119@mail.gmail.com>
	<191c3a030812021116n724f77c8oe477a1585f12e8da@mail.gmail.com>
Message-ID: <2d9149cd0812021120m2774d712qe95c9c32ecdfb85b@mail.gmail.com>

On 12/2/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> We'll schedule a round table with the topic
>
> SIP OMFG STFU
>
> At the next ClueCon aug 4th-6th 2009 to stir things up a bit =D
>

Heh.  I've been trying to make it back these last couple of years.  I
just might make it in '09!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From c_cav_01 at yahoo.com  Tue Dec  2 11:29:03 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 11:29:03 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20798791.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com>
	<B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>
	<20798791.post@talk.nabble.com>
Message-ID: <20799146.post@talk.nabble.com>


Note: while reading up on regex, I see that the ',' in ([0,1]) is superflous,
has been removed.  regex is now:
^([01]?)(8162565804)$
Didn't fix the problem but I'm a perfectionist, had to be changed. :D
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20799146.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From per_moeller at mac.com  Tue Dec  2 11:31:59 2008
From: per_moeller at mac.com (=?iso-8859-1?Q?Per_M=F8ller?=)
Date: Tue, 02 Dec 2008 20:31:59 +0100
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
References: <000001c9530d$912d86d0$b3889470$@com>
	<5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
Message-ID: <000f01c954b4$a616fa60$f244ef20$@com>

I checked out the trunk version, and it's still slow. However I found one
improvement - it does not crash on shutdown anymore.

Could anymore give me some pointers on how to try to debug this on the
Windows platform?


// Per

Fra: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Carlos
Talbot
Sendt: 2. december 2008 17:13
Til: freeswitch-users at lists.freeswitch.org
Emne: Re: [Freeswitch-users] Windows is slow?

Have you tried the latest msi build? It's based off svn 10564.

Carlos

On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:
I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
on a Windows Vista machine using the precompiled .msi - actually the same
machine).

Using the default configuration files, and using 2 Snom 360 phones I dialed
from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
instantly, but under Windows it takes 1-2 seconds before it starts ringing.

It seems to be in the dialplan the time is spent. From the time I see this
line on the console:

[INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
context default

Until the next thing happens it always takes at least 1 full second, but on
the Mac it happens instantly.

Why is the Windows build this much slower? Is it a known problem?

I get the feeling that the majority of the FS community is Unix based, which
is fine by me, but I would really like to know just how well supported and
stable the Win32 build is and if this is currently a viable way to go, or if
I should stick to Linux/BSD/Mac for production use?


// Per


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





From mgg at giagnocavo.net  Tue Dec  2 11:33:48 2008
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 2 Dec 2008 14:33:48 -0500
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <000f01c954b4$a616fa60$f244ef20$@com>
References: <000001c9530d$912d86d0$b3889470$@com>
	<5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
	<000f01c954b4$a616fa60$f244ef20$@com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702336DC03C@mse17be1.mse17.exchange.ms>

Can you do a console loglevel debug, then send all the output around that time?

Apart from that, the quickest way might just to attach a debugger, then break all when it pauses and see where the threads are :).

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Per M?ller
Sent: Tuesday, December 02, 2008 12:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Windows is slow?

I checked out the trunk version, and it's still slow. However I found one
improvement - it does not crash on shutdown anymore.

Could anymore give me some pointers on how to try to debug this on the
Windows platform?


// Per

Fra: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Carlos
Talbot
Sendt: 2. december 2008 17:13
Til: freeswitch-users at lists.freeswitch.org
Emne: Re: [Freeswitch-users] Windows is slow?

Have you tried the latest msi build? It's based off svn 10564.

Carlos

On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:
I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
on a Windows Vista machine using the precompiled .msi - actually the same
machine).

Using the default configuration files, and using 2 Snom 360 phones I dialed
from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
instantly, but under Windows it takes 1-2 seconds before it starts ringing.

It seems to be in the dialplan the time is spent. From the time I see this
line on the console:

[INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
context default

Until the next thing happens it always takes at least 1 full second, but on
the Mac it happens instantly.

Why is the Windows build this much slower? Is it a known problem?

I get the feeling that the majority of the FS community is Unix based, which
is fine by me, but I would really like to know just how well supported and
stable the Win32 build is and if this is currently a viable way to go, or if
I should stick to Linux/BSD/Mac for production use?


// Per


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From mrjoebain at gmail.com  Tue Dec  2 06:29:12 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 2 Dec 2008 14:29:12 +0000
Subject: [Freeswitch-users] Javascript ODBC on Windows
Message-ID: <748d53500812020629p6a0d178dh672cec871c018254@mail.gmail.com>

Hi all,

Is it possible to use mod_spidermonkey_odbc with a Windows installation of
FreeSWITCH at the moment? If so does anyone have any pointers? I get:

2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145 switch_odbc_handle_connect()
Connecting ivr_test
2008-12-02 14:23:57 [ERR] switch_odbc.c:160 switch_odbc_handle_connect()
STATE: IM002 CODE 0 ERROR: [Microsoft][ODBC Driver Manager] Data source name
not found and no default driver specified

when I try.

Thanks in advance,

Joe Bain
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From anthony.minessale at gmail.com  Tue Dec  2 12:05:15 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Dec 2008 14:05:15 -0600
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702336DC03C@mse17be1.mse17.exchange.ms>
References: <000001c9530d$912d86d0$b3889470$@com>
	<5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
	<000f01c954b4$a616fa60$f244ef20$@com>
	<6E8D2069C08AA84A83D336E996AE4C6702336DC03C@mse17be1.mse17.exchange.ms>
Message-ID: <191c3a030812021205r619ad735le129731ccb8f69d0@mail.gmail.com>

is it stun timeout ?

do you have one of the ip set to stun:foo ?


On Tue, Dec 2, 2008 at 1:33 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

> Can you do a console loglevel debug, then send all the output around that
> time?
>
> Apart from that, the quickest way might just to attach a debugger, then
> break all when it pauses and see where the threads are :).
>
> -Michael
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Per M?ller
> Sent: Tuesday, December 02, 2008 12:32 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Windows is slow?
>
> I checked out the trunk version, and it's still slow. However I found one
> improvement - it does not crash on shutdown anymore.
>
> Could anymore give me some pointers on how to try to debug this on the
> Windows platform?
>
>
> // Per
>
> Fra: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Carlos
> Talbot
> Sendt: 2. december 2008 17:13
> Til: freeswitch-users at lists.freeswitch.org
> Emne: Re: [Freeswitch-users] Windows is slow?
>
> Have you tried the latest msi build? It's based off svn 10564.
>
> Carlos
>
> On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:
> I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
> on a Windows Vista machine using the precompiled .msi - actually the same
> machine).
>
> Using the default configuration files, and using 2 Snom 360 phones I dialed
> from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
> instantly, but under Windows it takes 1-2 seconds before it starts ringing.
>
> It seems to be in the dialplan the time is spent. From the time I see this
> line on the console:
>
> [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
> context default
>
> Until the next thing happens it always takes at least 1 full second, but on
> the Mac it happens instantly.
>
> Why is the Windows build this much slower? Is it a known problem?
>
> I get the feeling that the majority of the FS community is Unix based,
> which
> is fine by me, but I would really like to know just how well supported and
> stable the Win32 build is and if this is currently a viable way to go, or
> if
> I should stick to Linux/BSD/Mac for production use?
>
>
> // Per
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Tue Dec  2 12:32:49 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 12:32:49 -0800
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
Message-ID: <87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>

Right now this page is up-to-date with the latest info:
http://wiki.freeswitch.org/wiki/Mod_fax

T.38 is not (yet) supported.

-MC

On Tue, Dec 2, 2008 at 9:40 AM, Dennis <odermann at googlemail.com> wrote:

> hi,
>
> because we do not get tired of testing and playing a lot with the
> beloved fs, we now arrived at the fax feature :-)
>
> i am not sure if the docs are up to date or if there was a lot of
> development in the meantime. therefore i would like to ask, what is
> possible and what will come in the near future.
>
> we are using fs, socket outbound and php and would like to make
> something like fax to mail as an additional service.
>
> is t38 supported?
> can i pass incoming faxes over the same socket as calls?
> can i convert faxes into pdf?
> is fax over sip reliable (as far as i have heard, under asterisk fax
> is nothing one should use)?
> and so on, and so on....
>
> i would be very happy to hear some user experiences with fs and fax.
> if it seems, that we can use fax with over socket outbound, we will do
> hardcore testing ;-)
>
> thanks,
> dennis
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kkielhofner at star2star.com  Tue Dec  2 13:28:28 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Tue, 2 Dec 2008 16:28:28 -0500
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
Message-ID: <2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>

On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <msc at freeswitch.org> wrote:
> Right now this page is up-to-date with the latest info:
> http://wiki.freeswitch.org/wiki/Mod_fax
>
> T.38 is not (yet) supported.
>
> -MC
>

Can you (or someone) elaborate on this?  Maybe the answer really is
no, but what about support for UDPTL, pass through, etc?

It looks like Sofia should be good to go...

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From mike at jerris.com  Tue Dec  2 14:36:34 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 17:36:34 -0500
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
Message-ID: <40890CF2-2279-464A-A58D-A86087D2CD1A@jerris.com>

T.38 passthrough IS supported, T.38 endpoint and gateway are not yet  
supported.

Mike

On Dec 2, 2008, at 4:28 PM, Kristian Kielhofner wrote:

> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <msc at freeswitch.org>  
> wrote:
>> Right now this page is up-to-date with the latest info:
>> http://wiki.freeswitch.org/wiki/Mod_fax
>>
>> T.38 is not (yet) supported.
>>
>> -MC
>>
>
> Can you (or someone) elaborate on this?  Maybe the answer really is
> no, but what about support for UDPTL, pass through, etc?
>
> It looks like Sofia should be good to go...
>



From mike at jerris.com  Tue Dec  2 14:38:40 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Dec 2008 17:38:40 -0500
Subject: [Freeswitch-users] Javascript ODBC on Windows
In-Reply-To: <748d53500812020629p6a0d178dh672cec871c018254@mail.gmail.com>
References: <748d53500812020629p6a0d178dh672cec871c018254@mail.gmail.com>
Message-ID: <E05E9383-0BF2-4B8B-93A5-09FB99817F19@jerris.com>

Yes, it should work fine.  As the error message says it didn't find  
the data source name you specified.  You need to setup your odbc data  
source on the system

Mike

On Dec 2, 2008, at 9:29 AM, Joe Bain wrote:

> Hi all,
>
> Is it possible to use mod_spidermonkey_odbc with a Windows  
> installation of FreeSWITCH at the moment? If so does anyone have any  
> pointers? I get:
>
> 2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145  
> switch_odbc_handle_connect() Connecting ivr_test
> 2008-12-02 14:23:57 [ERR] switch_odbc.c:160  
> switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [Microsoft] 
> [ODBC Driver Manager] Data source name not found and no default  
> driver specified
>
> when I try.
>
> Thanks in advance,
>
> Joe Bain
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Tue Dec  2 14:39:06 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2008 16:39:06 -0600
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20799146.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com>
	<B22005DB-5F5F-45C2-B269-355DF309D578@freeswitch.org>
	<20798791.post@talk.nabble.com> <20799146.post@talk.nabble.com>
Message-ID: <AF0CB43E-C3D1-442B-A29F-0DBE21E71025@freeswitch.org>

After you set ${dialed_user}=$2 try using ${dialed_user} everywhere  
instead of $2 just to test.

/b

On Dec 2, 2008, at 1:29 PM, ccav wrote:

>
> Note: while reading up on regex, I see that the ',' in ([0,1]) is  
> superflous,
> has been removed.  regex is now:
> ^([01]?)(8162565804)$
> Didn't fix the problem but I'm a perfectionist, had to be changed. :D
> -- 
> View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20799146.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From c_cav_01 at yahoo.com  Tue Dec  2 16:05:47 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 16:05:47 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20791453.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com>
Message-ID: <20803931.post@talk.nabble.com>


Made the change, no joy.

Do I need to set sip_req_user to the updated DID?

Also, I misspoke in my first post, apparently the bridge is NOT going
through either.  Is there some var/param I can set with $2 so I can see it
in the "info"?
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20803931.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From c_cav_01 at yahoo.com  Tue Dec  2 16:33:38 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 16:33:38 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20803931.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com> <20803931.post@talk.nabble.com>
Message-ID: <20804247.post@talk.nabble.com>


Okay, I found out who the culprit is, but I still want to find a fix so the
dialplan works like I want.

The <application="playback" causes an answer and early media...  When I
comment that out, the voicemail works almost properly.  It's clipping the
(81) off the front of the 816 DID, which is probably thinking those are pbx
dialout values.  If anyone can tell me where that is set so I can change
it...  But the bridge and voicemail both work like they are sposed to.

However, something is happening in playback that is making it impossible to
bridge the call or send it to voicemail properly.

-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804247.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From c_cav_01 at yahoo.com  Tue Dec  2 16:35:59 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 16:35:59 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
Message-ID: <20804247.post@talk.nabble.com>


Okay, I found out who the culprit is, but I still want to find a fix so the
dialplan works like I want.

The <application="playback" causes an answer and early media...  When I
comment that out, the voicemail works almost properly.  It's clipping the
(81) off the front of the 816 DID, which is probably thinking those are pbx
dialout values.  If anyone can tell me where that is set so I can change
it...  But the bridge and voicemail both work like they are sposed to.

However, something is happening in playback that is making it impossible to
bridge the call or send it to voicemail properly.

Instead of calling application playback and then trying to bridge, should I
just bridge to both?  What I want is the caller to hear my music instead of
the ring, and maybe this is the wrong way to do it?
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804247.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From msc at freeswitch.org  Tue Dec  2 16:48:10 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 16:48:10 -0800
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
Message-ID: <87f2f3b90812021648v7f1402ddma747ea0da1eac577@mail.gmail.com>

On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Right now this page is up-to-date with the latest info:
> > http://wiki.freeswitch.org/wiki/Mod_fax
> >
> > T.38 is not (yet) supported.
> >
> > -MC
> >
>
> Can you (or someone) elaborate on this?  Maybe the answer really is
> no, but what about support for UDPTL, pass through, etc?
>

Excellent questions! I will research and report back to the list...
-MC


>
> It looks like Sofia should be good to go...
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From c_cav_01 at yahoo.com  Tue Dec  2 17:11:52 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 2 Dec 2008 17:11:52 -0800 (PST)
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20804247.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com> <20803931.post@talk.nabble.com>
	<20804247.post@talk.nabble.com>
Message-ID: <20804652.post@talk.nabble.com>


RESOLVED.

Duh, I'm sposed to use ringback, not playback...

Someone should write a book on this...  Maybe I will.
-- 
View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804652.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From msc at freeswitch.org  Tue Dec  2 17:18:22 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 17:18:22 -0800
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
Message-ID: <87f2f3b90812021718j5aae69aav6dd8ee7953e2b1ff@mail.gmail.com>

Kristian,

Are you on the IRC channel by any chance?
-MC (IRC: mercutioviz)

On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner <
kkielhofner at star2star.com> wrote:

> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Right now this page is up-to-date with the latest info:
> > http://wiki.freeswitch.org/wiki/Mod_fax
> >
> > T.38 is not (yet) supported.
> >
> > -MC
> >
>
> Can you (or someone) elaborate on this?  Maybe the answer really is
> no, but what about support for UDPTL, pass through, etc?
>
> It looks like Sofia should be good to go...
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Dec  2 17:18:53 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 17:18:53 -0800
Subject: [Freeswitch-users] Wrong # in voicemail
In-Reply-To: <20804652.post@talk.nabble.com>
References: <20791453.post@talk.nabble.com> <20803931.post@talk.nabble.com>
	<20804247.post@talk.nabble.com> <20804652.post@talk.nabble.com>
Message-ID: <87f2f3b90812021718k1b0c51dcx8740e3c387ce4887@mail.gmail.com>

hehe, careful what you wish for...

On Tue, Dec 2, 2008 at 5:11 PM, ccav <c_cav_01 at yahoo.com> wrote:

>
> RESOLVED.
>
> Duh, I'm sposed to use ringback, not playback...
>
> Someone should write a book on this...  Maybe I will.
> --
> View this message in context:
> http://www.nabble.com/Wrong---in-voicemail-tp20791453p20804652.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From klaus.teller at gmx.net  Tue Dec  2 19:15:52 2008
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 03 Dec 2008 04:15:52 +0100
Subject: [Freeswitch-users] Bridging from Event Socket API
Message-ID: <20081203031552.178560@gmx.net>

Hi Folks,

so far i could understand how to bridge calls with Javascript. I'm trying to do the same with Java via the Socket Interface. My first trials weren't successful. maybe you can help me understand what is goin on. 

What i want to do is to bridge an existing leg (Unique-ID is known) to a party that wasn't yet dialed (Unique-ID unknown). With javascript it is something like:

session.bridge("sofia/internal/1002");

How do i do this using the event socket interface? what application/command would i use with which arguments?


One way i tried to do this is to orginate a call to 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't successful. The only message i had on the FS console is: 

2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693 switch_core_session_queue_private_event() Send signal sofia/internal/1001 at 192.168.1.121 [BREAK]

Any idea what i'm missing?

Thanks,

Klaus.





-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From msc at freeswitch.org  Tue Dec  2 20:39:08 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2008 20:39:08 -0800
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <20081203031552.178560@gmx.net>
References: <20081203031552.178560@gmx.net>
Message-ID: <87f2f3b90812022039odcacf9dte67c33707e41efc0@mail.gmail.com>

You probably have several options depending upon your needs. Could you
elaborate a bit on what the big picture is? Also, what exactly were you
doing when you established the second call leg? Did the second call let get
created and a valid uuid assigned, etc.? Just checking.

Let us know,
MC

On Tue, Dec 2, 2008 at 7:15 PM, Klaus Teller <klaus.teller at gmx.net> wrote:

> Hi Folks,
>
> so far i could understand how to bridge calls with Javascript. I'm trying
> to do the same with Java via the Socket Interface. My first trials weren't
> successful. maybe you can help me understand what is goin on.
>
> What i want to do is to bridge an existing leg (Unique-ID is known) to a
> party that wasn't yet dialed (Unique-ID unknown). With javascript it is
> something like:
>
> session.bridge("sofia/internal/1002");
>
> How do i do this using the event socket interface? what application/command
> would i use with which arguments?
>
>
> One way i tried to do this is to orginate a call to 'sofia/internal/1002'
> and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't
> successful. The only message i had on the FS console is:
>
> 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693
> switch_core_session_queue_private_event() Send signal sofia/internal/
> 1001 at 192.168.1.121 [BREAK]
>
> Any idea what i'm missing?
>
> Thanks,
>
> Klaus.
>
>
>
>
>
> --
> Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen:
> http://www.gmx.net/de/go/multimessenger
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dave at 3c.co.uk  Wed Dec  3 03:15:57 2008
From: dave at 3c.co.uk (David Knell)
Date: Wed, 03 Dec 2008 11:15:57 +0000
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <20081203031552.178560@gmx.net>
References: <20081203031552.178560@gmx.net>
Message-ID: <49366A6D.4000702@3c.co.uk>

Hi Klaus,

Some Perl code snippets - we use:
call_command("bridge", "sofia/gateway/bt/$ntd");
which, in turn, is:
sub call_command($$) {
        my $cmd = shift;
        my $arg = shift;
        print $sock "sendmsg\ncall-command: execute\nexecute-app-name: 
$cmd\nexecute-app-arg: $arg\n\n";
}

Cheers --

Dave

> Hi Folks,
>
> so far i could understand how to bridge calls with Javascript. I'm trying to do the same with Java via the Socket Interface. My first trials weren't successful. maybe you can help me understand what is goin on. 
>
> What i want to do is to bridge an existing leg (Unique-ID is known) to a party that wasn't yet dialed (Unique-ID unknown). With javascript it is something like:
>
> session.bridge("sofia/internal/1002");
>
> How do i do this using the event socket interface? what application/command would i use with which arguments?
>
>
> One way i tried to do this is to orginate a call to 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't successful. The only message i had on the FS console is: 
>
> 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693 switch_core_session_queue_private_event() Send signal sofia/internal/1001 at 192.168.1.121 [BREAK]
>
> Any idea what i'm missing?
>
> Thanks,
>
> Klaus.
>
>
>
>
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 




From regs at kinetix.gr  Wed Dec  3 05:30:00 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 15:30:00 +0200
Subject: [Freeswitch-users] How to get info from the b-leg
Message-ID: <493689D8.9040708@kinetix.gr>

Hi,

    I am making a simple bridge between two call legs :

Client --(a-leg)--> FS --(b-leg)-->Provider

How can I get information like network-address of the Provider, 
media-address,
port used, media port used etc. from the second leg (b-leg)?

Is all the information provided by the a-leg available for the b-leg as 
well? If, yes
how can I access it? (and log it to my CDR file eventually)



From anthony.minessale at gmail.com  Wed Dec  3 05:53:22 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 07:53:22 -0600
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <493689D8.9040708@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
Message-ID: <191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>

2 options.
1) enable b-leg logging on the cdr module.
2) you can use the prefix bleg_ in a variable context to get to
caller_profile members
from the b leg.

eg ${bleg_caller_id_name}


On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:

> Hi,
>
>    I am making a simple bridge between two call legs :
>
> Client --(a-leg)--> FS --(b-leg)-->Provider
>
> How can I get information like network-address of the Provider,
> media-address,
> port used, media port used etc. from the second leg (b-leg)?
>
> Is all the information provided by the a-leg available for the b-leg as
> well? If, yese
> how can I access it? (and log it to my CDR file eventually)
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From regs at kinetix.gr  Wed Dec  3 06:18:31 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 16:18:31 +0200
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
Message-ID: <49369537.6040008@kinetix.gr>

b-leg logging is enabled in the cdr module. but in the cdrs I cannot get 
any variables that refer to the b-leg.

I tried the second way using ${sip_to_host} and {bleg_sip_to_host} but :

a) the variable returns the FS IP on the a-leg CDR (correctly)
b) the variable returns nothing on the b-leg CDR (empty). Shouldn't it 
return the "to" host of the b-leg (my providers address)?


Anthony Minessale wrote:
> 2 options.
> 1) enable b-leg logging on the cdr module.
> 2) you can use the prefix bleg_ in a variable context to get to 
> caller_profile members
> from the b leg.
>
> eg ${bleg_caller_id_name}
>
>
> On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr 
> <mailto:regs at kinetix.gr> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     Hi,
>
>        I am making a simple bridge between two call legs :
>
>     Client --(a-leg)--> FS --(b-leg)-->Provider
>
>     How can I get information like network-address of the Provider,
>     media-address,
>     port used, media port used etc. from the second leg (b-leg)?
>
>     Is all the information provided by the a-leg available for the
>     b-leg as
>     well? If, yese
>     how can I access it? (and log it to my CDR file eventually)
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From r.pankratz at fh-wolfenbuettel.de  Wed Dec  3 06:21:16 2008
From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz)
Date: Wed, 03 Dec 2008 15:21:16 +0100
Subject: [Freeswitch-users] Dialing tone when placing a call
	with	portaudio
In-Reply-To: <ADE59C30-1DBE-4DB5-B96B-9EDA0A0DCE07@jerris.com>
References: <20081120154859.16192.qmail@server15.citromail.hu>
	<4934FBED.7030307@fh-wolfenbuettel.de>
	<ADE59C30-1DBE-4DB5-B96B-9EDA0A0DCE07@jerris.com>
Message-ID: <493695DC.4060706@fh-wolfenbuettel.de>

You were right, I was calling sip when not getting a dialing tone.
In the SIP flow I get a 180 ringing and no 183.
Setting ringback channel fixed that issue.

Thanks for your help!

Ren?
> What are you calling, sip I assume, this may be a case where the sip  
> signaling is sending a 180 ringing instead of a 183 and we are not  
> generating ringback in that case.  Can you please confirm that and  
> test if setting the ringback channel variable before bridge fixes this  
> issue?
>
> Mike
>
> On Dec 2, 2008, at 4:12 AM, Rene Pankratz wrote:
>
>   
>> Hello,
>> when using mod_portaudio for calling somebody I don't hear anything
>> until the other party answers the call. Is it possible to play a  
>> dialing
>> tone when the other party is ringing?
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   





From klaus.teller at gmx.net  Wed Dec  3 06:48:02 2008
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 03 Dec 2008 15:48:02 +0100
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <49366A6D.4000702@3c.co.uk>
References: <20081203031552.178560@gmx.net> <49366A6D.4000702@3c.co.uk>
Message-ID: <20081203144802.281260@gmx.net>

Hi All,

Thanks for your feedback. I must be doing something fundamentally wrong. Inbound socket is working without problems. But the exact things that i do on inbound socket, i'm not able to replcate them on outbound socket.

The global picture: I have on Xlite registered at extension 1002 and another one at extension 1003.  Then i have an extension 8998 in the default context. Here is the extension definition:

   <extension name="outbound-test">
        <condition field="destination_number" expression="^8998$" >
            <action application="socket" data="192.168.50.56:10000 full" />
        </condition>		
    </extension>

I use Xlite-1003 to call this extension (8998) and the call is properly notified to the remote Java server. 

Then on the Java side, after receiving the event, i send a CONNECT command: "Connect\n\n"
The answer from Freeswitch is the state of the channel ( a set of variable, value pair).

Up to this point everything seems normal to me. But then, i try to send an answer command:

sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
call-command: execute
execute-app-name: answer
execute-app-arg:

Freswitch replies with: 

Content-Type: command/reply
Reply-Text: +OK

But the call is still not answered. Nothing happens on the freeswitch console (Log level DEBUG) and the dialing XLite is still in calling modus.

Then i try bridging the call to 1002:

sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
call-command: execute
execute-app-name: bridge
execute-app-arg: sofia/internal/1002%192.168.50.94
   
Again Freeswitch does answer with:

Content-Type: command/reply
Reply-Text: +OK

And yet again, nothing is really happening.

What am i missing here?

Thanks,
Klaus.

-------- Original-Nachricht --------
> Datum: Wed, 03 Dec 2008 11:15:57 +0000
> Von: David Knell <dave at 3c.co.uk>
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Bridging from Event Socket API

> Hi Klaus,
> 
> Some Perl code snippets - we use:
> call_command("bridge", "sofia/gateway/bt/$ntd");
> which, in turn, is:
> sub call_command($$) {
>         my $cmd = shift;
>         my $arg = shift;
>         print $sock "sendmsg\ncall-command: execute\nexecute-app-name: 
> $cmd\nexecute-app-arg: $arg\n\n";
> }
> 
> Cheers --
> 
> Dave
> 
> > Hi Folks,
> >
> > so far i could understand how to bridge calls with Javascript. I'm
> trying to do the same with Java via the Socket Interface. My first trials
> weren't successful. maybe you can help me understand what is goin on. 
> >
> > What i want to do is to bridge an existing leg (Unique-ID is known) to a
> party that wasn't yet dialed (Unique-ID unknown). With javascript it is
> something like:
> >
> > session.bridge("sofia/internal/1002");
> >
> > How do i do this using the event socket interface? what
> application/command would i use with which arguments?
> >
> >
> > One way i tried to do this is to orginate a call to
> 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it
> wasn't successful. The only message i had on the FS console is: 
> >
> > 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693
> switch_core_session_queue_private_event() Send signal sofia/internal/1001 at 192.168.1.121
> [BREAK]
> >
> > Any idea what i'm missing?
> >
> > Thanks,
> >
> > Klaus.
> >
> >
> >
> >
> >
> >   
> 
> 
> -- 
> David Knell, Director, 3C Limited
> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
> http://www.3c.co.uk 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From yudha2008 at gmail.com  Wed Dec  3 07:32:12 2008
From: yudha2008 at gmail.com (Baskar)
Date: Wed, 3 Dec 2008 21:02:12 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
Message-ID: <d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>

Hi,

*I have newly installed freeswitch in another machine.

**After starting the freeswitch I try to dial a extension from console but
when i call extension 1002 from freeswitch console, call is connected to
extension 1002, but FS is aborted but call is established in1002.*

*When i dial from console the call get connected and freeswitch is cut.*

*OUtput:*


*FreeSWITCH Version 1.0.trunk (10567) Started.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
2008-12-03 21:02:21 [CONSOLE] mod_local_stream.c:142 read_stream_thread()
Can't open directory: /usr/local/freeswitch/sounds/music/16000*
*
freeswitch at hp30094686650.optimus.co.in> pa devlist*
*API CALL [pa(devlist)] output:
0;/dev/dsp;16;4
1;Intel ICH5: Intel ICH5 (hw:0,0);2;6
2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0
3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0
4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0
5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2
6;front;0;6
7;surround40;0;4
8;surround41;0;128
9;surround50;0;128
10;surround51;0;6
11;iec958;0;2
12;spdif;0;2
13;default;128;128
14;dmix;0;2*

*freeswitch at hp30094686650.optimus.co.in> pa call 1002*
*2008-12-03 21:06:11 [NOTICE] switch_channel.c:564 switch_channel_set_name()
New Channel portaudio/1002 [fae97d5b-3480-410e-af0a-192d00710537]
freeswitch at hp30094686650.optimus.co.in> 2008-12-03 21:06:12 [NOTICE]
mod_portaudio.c:1586 place_call() Channel [portaudio/1002] has been answered
API CALL [pa(call 1002)] output:
SUCCESS:1:fae97d5b-3480-410e-af0a-192d00710537

2008-12-03 21:06:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
FreeSWITCH->1002 in context default
2008-12-03 21:06:12 [WARNING] switch_ivr.c:1840 switch_ivr_set_user() can't
find user [default@]
2008-12-03 21:06:12 [INFO] mod_dptools.c:872 info_function() CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [portaudio/1002]
Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537]
Call-Direction: [inbound]
Answer-State: [answered]
Channel-Read-Codec-Name: [L16]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [L16]
Channel-Write-Codec-Rate: [8000]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [FreeSWITCH]
Caller-Caller-ID-Number: [0000000000]
Caller-Network-Addr: [172.20.176.32]
Caller-Destination-Number: [1002]
Caller-Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537]
Caller-Source: [mod_portaudio]
Caller-Context: [default]
Caller-Channel-Name: [portaudio/1002]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1228318571584600]
Caller-Channel-Created-Time: [1228318571584600]
Caller-Channel-Answered-Time: [1228318572164620]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_channel_name: [portaudio/1002]
variable_endpoint_disposition: [ANSWER]
variable_read_codec: [L16]
variable_read_rate: [8000]
variable_write_codec: [L16]
variable_write_rate: [8000]
variable_use_profile: [nat]
variable_dialed_ext: [1002]
variable_current_application: [info]


2008-12-03 21:06:12 [INFO] mod_dptools.c:858 log_function() Answer-State []n
2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML
features
2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2
record_session::/usr/local/freeswitch/recordings/0000000000.2008-12-03-21-06-12.wav
2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML
features
2008-12-03 21:06:12 [NOTICE] switch_channel.c:564 switch_channel_set_name()
New Channel sofia/internal/sip:1002 at 172.20.179.201:37046;rinstance=e6259d34a17a130a
[e9a905cd-dc7c-49b1-b3f7-1cd52c1129d1]*
*freeswitch: src/switch_core_io.c:202: switch_core_session_read_frame:
Assertion `(*frame)->codec != ((void *)0)' failed.
Aborted (core dumped)
[root at hp30094686650 bin]# *
*

After installing current svn trunk also i get the same error.I cant able to
recover the failure .Correct me were i am wrong.


Thanks Regards,
N.Baskar
*
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From anthony.minessale at gmail.com  Wed Dec  3 07:37:07 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 09:37:07 -0600
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <49369537.6040008@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
	<49369537.6040008@kinetix.gr>
Message-ID: <191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>

outgoing calls to not have an ip value set.
if you want to store the dest ip in the cdr you need to set it as a custom
variable and insert it
into your template for csv cdr or it will just be there in xml cdr

On Wed, Dec 3, 2008 at 8:18 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:

>  b-leg logging is enabled in the cdr module. but in the cdrs I cannot get
> any variables that refer to the b-leg.
>
> I tried the second way using ${sip_to_host} and {bleg_sip_to_host} but :
>
> a) the variable returns the FS IP on the a-leg CDR (correctly)
> b) the variable returns nothing on the b-leg CDR (empty). Shouldn't it
> return the "to" host of the b-leg (my providers address)?
>
>
> Anthony Minessale wrote:
>
> 2 options.
> 1) enable b-leg logging on the cdr module.
> 2) you can use the prefix bleg_ in a variable context to get to
> caller_profile members
> from the b leg.
>
> eg ${bleg_caller_id_name}
>
>
> On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:
>
>> Hi,
>>
>>    I am making a simple bridge between two call legs :
>>
>> Client --(a-leg)--> FS --(b-leg)-->Provider
>>
>> How can I get information like network-address of the Provider,
>> media-address,
>> port used, media port used etc. from the second leg (b-leg)?
>>
>> Is all the information provided by the a-leg available for the b-leg as
>> well? If, yese
>> how can I access it? (and log it to my CDR file eventually)
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From regs at kinetix.gr  Wed Dec  3 07:48:03 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 17:48:03 +0200
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
References: <493689D8.9040708@kinetix.gr>	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>	<49369537.6040008@kinetix.gr>
	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
Message-ID: <4936AA33.7080301@kinetix.gr>

I looked in the b-leg xml cdr and the ip address is not there (for 
signaling) it is only there
for media (${remote_media_ip}) which is not the same thing now, is it?

While we are at it, I noticed that the ${local_media_port} and 
${remote_media_port}
have the same value for each CDR (a or b leg). Shouldn't the first 
variable hold the port
of the FS (on both legs) and the second variable the port of the client 
(in the a-leg) or the port of
the provider (in the b-leg)?

Anthony Minessale wrote:
> outgoing calls to not have an ip value set.
> if you want to store the dest ip in the cdr you need to set it as a 
> custom variable and insert it
> into your template for csv cdr or it will just be there in xml cdr
>
> On Wed, Dec 3, 2008 at 8:18 AM, regs at kinetix.gr 
> <mailto:regs at kinetix.gr> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     b-leg logging is enabled in the cdr module. but in the cdrs I
>     cannot get any variables that refer to the b-leg.
>
>     I tried the second way using ${sip_to_host} and {bleg_sip_to_host}
>     but :
>
>     a) the variable returns the FS IP on the a-leg CDR (correctly)
>     b) the variable returns nothing on the b-leg CDR (empty).
>     Shouldn't it return the "to" host of the b-leg (my providers address)?
>
>
>     Anthony Minessale wrote:
>>     2 options.
>>     1) enable b-leg logging on the cdr module.
>>     2) you can use the prefix bleg_ in a variable context to get to
>>     caller_profile members
>>     from the b leg.
>>
>>     eg ${bleg_caller_id_name}
>>
>>
>>     On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr
>>     <mailto:regs at kinetix.gr> <regs at kinetix.gr
>>     <mailto:regs at kinetix.gr>> wrote:
>>
>>         Hi,
>>
>>            I am making a simple bridge between two call legs :
>>
>>         Client --(a-leg)--> FS --(b-leg)-->Provider
>>
>>         How can I get information like network-address of the Provider,
>>         media-address,
>>         port used, media port used etc. from the second leg (b-leg)?
>>
>>         Is all the information provided by the a-leg available for
>>         the b-leg as
>>         well? If, yese
>>         how can I access it? (and log it to my CDR file eventually)
>>
>>         _______________________________________________
>>         Freeswitch-users mailing list
>>         Freeswitch-users at lists.freeswitch.org
>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Anthony Minessale II
>>
>>     FreeSWITCH http://www.freeswitch.org/
>>     ClueCon http://www.cluecon.com/
>>
>>     AIM: anthm
>>     MSN:anthony_minessale at hotmail.com
>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>>     FreeSWITCH Developer Conference
>>     sip:888 at conference.freeswitch.org
>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>     iax:guest at conference.freeswitch.org/888
>>     <http://iax:guest at conference.freeswitch.org/888>
>>     googletalk:conf+888 at conference.freeswitch.org
>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>     pstn:213-799-1400
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From msc at freeswitch.org  Wed Dec  3 08:04:35 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 3 Dec 2008 08:04:35 -0800
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<7b197bef0812010315l2866875bscb757f0aacd762ac@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
	<d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
Message-ID: <3198D996-3558-4539-A1E4-1D3C09835388@freeswitch.org>

Baskar,

Which operating system are you running? I would like to try and  
duplicate symptoms on one of my boxes, all of which run CentOS 5.x

-MC

Sent from my iPhone

On Dec 3, 2008, at 7:32 AM, Baskar <yudha2008 at gmail.com> wrote:

> Hi,
>
> I have newly installed freeswitch in another machine.
>
> After starting the freeswitch I try to dial a extension from console  
> but when i call extension 1002 from freeswitch console, call is  
> connected to extension 1002, but FS is aborted but call is  
> established in1002.
>
> When i dial from console the call get connected and freeswitch is cut.
>
> OUtput:
>
>
> FreeSWITCH Version 1.0.trunk (10567) Started.
> Crash Protection [Disabled]
> Max Sessions[1000]
> Session Rate[30]
> SQL [Enabled]
> 2008-12-03 21:02:21 [CONSOLE] mod_local_stream.c:142  
> read_stream_thread() Can't open directory: /usr/local/freeswitch/ 
> sounds/music/16000
>
> freeswitch at hp30094686650.optimus.co.in> pa devlist
> API CALL [pa(devlist)] output:
> 0;/dev/dsp;16;4
> 1;Intel ICH5: Intel ICH5 (hw:0,0);2;6
> 2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0
> 3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0
> 4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0
> 5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2
> 6;front;0;6
> 7;surround40;0;4
> 8;surround41;0;128
> 9;surround50;0;128
> 10;surround51;0;6
> 11;iec958;0;2
> 12;spdif;0;2
> 13;default;128;128
> 14;dmix;0;2
>
> freeswitch at hp30094686650.optimus.co.in> pa call 1002
> 2008-12-03 21:06:11 [NOTICE] switch_channel.c:564  
> switch_channel_set_name() New Channel portaudio/1002  
> [fae97d5b-3480-410e-af0a-192d00710537]
> freeswitch at hp30094686650.optimus.co.in> 2008-12-03 21:06:12 [NOTICE]  
> mod_portaudio.c:1586 place_call() Channel [portaudio/1002] has been  
> answered
> API CALL [pa(call 1002)] output:
> SUCCESS:1:fae97d5b-3480-410e-af0a-192d00710537
>
> 2008-12-03 21:06:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()  
> Processing FreeSWITCH->1002 in context default
> 2008-12-03 21:06:12 [WARNING] switch_ivr.c:1840  
> switch_ivr_set_user() can't find user [default@]
> 2008-12-03 21:06:12 [INFO] mod_dptools.c:872 info_function()  
> CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [portaudio/1002]
> Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537]
> Call-Direction: [inbound]
> Answer-State: [answered]
> Channel-Read-Codec-Name: [L16]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [L16]
> Channel-Write-Codec-Rate: [8000]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [FreeSWITCH]
> Caller-Caller-ID-Number: [0000000000]
> Caller-Network-Addr: [172.20.176.32]
> Caller-Destination-Number: [1002]
> Caller-Unique-ID: [fae97d5b-3480-410e-af0a-192d00710537]
> Caller-Source: [mod_portaudio]
> Caller-Context: [default]
> Caller-Channel-Name: [portaudio/1002]
> Caller-Profile-Index: [1]
> Caller-Profile-Created-Time: [1228318571584600]
> Caller-Channel-Created-Time: [1228318571584600]
> Caller-Channel-Answered-Time: [1228318572164620]
> Caller-Channel-Progress-Time: [0]
> Caller-Channel-Progress-Media-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [true]
> Caller-Privacy-Hide-Name: [false]
> Caller-Privacy-Hide-Number: [false]
> variable_channel_name: [portaudio/1002]
> variable_endpoint_disposition: [ANSWER]
> variable_read_codec: [L16]
> variable_read_rate: [8000]
> variable_write_codec: [L16]
> variable_write_rate: [8000]
> variable_use_profile: [nat]
> variable_dialed_ext: [1002]
> variable_current_application: [info]
>
>
> 2008-12-03 21:06:12 [INFO] mod_dptools.c:858 log_function() Answer- 
> State []n
> 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1  
> execute_extension::dx XML features
> 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ 
> usr/local/freeswitch/recordings/0000000000.2008-12-03-21-06-12.wav
> 2008-12-03 21:06:12 [INFO] switch_ivr_async.c:1577  
> switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3  
> execute_extension::cf XML features
> 2008-12-03 21:06:12 [NOTICE] switch_channel.c:564  
> switch_channel_set_name() New Channel sofia/internal/sip:1002 at 172.20.179.201 
> :37046;rinstance=e6259d34a17a130a [e9a905cd-dc7c-49b1- 
> b3f7-1cd52c1129d1]
> freeswitch: src/switch_core_io.c:202:  
> switch_core_session_read_frame: Assertion `(*frame)->codec != ((void  
> *)0)' failed.
> Aborted (core dumped)
> [root at hp30094686650 bin]#
>
>
> After installing current svn trunk also i get the same error.I cant  
> able to recover the failure .Correct me were i am wrong.
>
>
> Thanks Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dave at 3c.co.uk  Wed Dec  3 08:21:47 2008
From: dave at 3c.co.uk (David Knell)
Date: Wed, 03 Dec 2008 16:21:47 +0000
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <20081203144802.281260@gmx.net>
References: <20081203031552.178560@gmx.net> <49366A6D.4000702@3c.co.uk>
	<20081203144802.281260@gmx.net>
Message-ID: <4936B21B.1030607@3c.co.uk>

Hi Klaus,

There's two differences that I can see between what you're doing and 
what we do:-
1.  We're using the socket in async mode (shouldn't make any difference)
2.  You don't need to send the UUID in after the sendmsg - FS already 
knows which call you're controlling.

Cheers --

Dave
> Hi All,
>
> Thanks for your feedback. I must be doing something fundamentally wrong. Inbound socket is working without problems. But the exact things that i do on inbound socket, i'm not able to replcate them on outbound socket.
>
> The global picture: I have on Xlite registered at extension 1002 and another one at extension 1003.  Then i have an extension 8998 in the default context. Here is the extension definition:
>
>    <extension name="outbound-test">
>         <condition field="destination_number" expression="^8998$" >
>             <action application="socket" data="192.168.50.56:10000 full" />
>         </condition>		
>     </extension>
>
> I use Xlite-1003 to call this extension (8998) and the call is properly notified to the remote Java server. 
>
> Then on the Java side, after receiving the event, i send a CONNECT command: "Connect\n\n"
> The answer from Freeswitch is the state of the channel ( a set of variable, value pair).
>
> Up to this point everything seems normal to me. But then, i try to send an answer command:
>
> sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
> call-command: execute
> execute-app-name: answer
> execute-app-arg:
>
> Freswitch replies with: 
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> But the call is still not answered. Nothing happens on the freeswitch console (Log level DEBUG) and the dialing XLite is still in calling modus.
>
> Then i try bridging the call to 1002:
>
> sendmsg  b30a2d2e-c146-11dd-9b99-07347b46e4ea
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: sofia/internal/1002%192.168.50.94
>    
> Again Freeswitch does answer with:
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> And yet again, nothing is really happening.
>
> What am i missing here?
>
> Thanks,
> Klaus.
>
> -------- Original-Nachricht --------
>   
>> Datum: Wed, 03 Dec 2008 11:15:57 +0000
>> Von: David Knell <dave at 3c.co.uk>
>> An: freeswitch-users at lists.freeswitch.org
>> Betreff: Re: [Freeswitch-users] Bridging from Event Socket API
>>     
>
>   
>> Hi Klaus,
>>
>> Some Perl code snippets - we use:
>> call_command("bridge", "sofia/gateway/bt/$ntd");
>> which, in turn, is:
>> sub call_command($$) {
>>         my $cmd = shift;
>>         my $arg = shift;
>>         print $sock "sendmsg\ncall-command: execute\nexecute-app-name: 
>> $cmd\nexecute-app-arg: $arg\n\n";
>> }
>>
>> Cheers --
>>
>> Dave
>>
>>     
>>> Hi Folks,
>>>
>>> so far i could understand how to bridge calls with Javascript. I'm
>>>       
>> trying to do the same with Java via the Socket Interface. My first trials
>> weren't successful. maybe you can help me understand what is goin on. 
>>     
>>> What i want to do is to bridge an existing leg (Unique-ID is known) to a
>>>       
>> party that wasn't yet dialed (Unique-ID unknown). With javascript it is
>> something like:
>>     
>>> session.bridge("sofia/internal/1002");
>>>
>>> How do i do this using the event socket interface? what
>>>       
>> application/command would i use with which arguments?
>>     
>>> One way i tried to do this is to orginate a call to
>>>       
>> 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it
>> wasn't successful. The only message i had on the FS console is: 
>>     
>>> 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693
>>>       
>> switch_core_session_queue_private_event() Send signal sofia/internal/1001 at 192.168.1.121
>> [BREAK]
>>     
>>> Any idea what i'm missing?
>>>
>>> Thanks,
>>>
>>> Klaus.
>>>
>>>
>>>
>>>
>>>
>>>   
>>>       
>> -- 
>> David Knell, Director, 3C Limited
>> T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
>> http://www.3c.co.uk 
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From klaus.teller at gmx.net  Wed Dec  3 08:43:37 2008
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 03 Dec 2008 17:43:37 +0100
Subject: [Freeswitch-users] Bridging from Event Socket API
In-Reply-To: <4936B21B.1030607@3c.co.uk>
References: <20081203031552.178560@gmx.net> <49366A6D.4000702@3c.co.uk>
	<20081203144802.281260@gmx.net> <4936B21B.1030607@3c.co.uk>
Message-ID: <20081203164337.63380@gmx.net>


> 
> 2.  You don't need to send the UUID in after the sendmsg - FS already 
> knows which call you're controlling.


Bingo! That was it.

Thanks,
Klaus.
-- 
Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL 
f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a



From anthony.minessale at gmail.com  Wed Dec  3 09:13:02 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 11:13:02 -0600
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <4936AA33.7080301@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
	<49369537.6040008@kinetix.gr>
	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
	<4936AA33.7080301@kinetix.gr>
Message-ID: <191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>

It's not an unreasonabe request so i added a patch you can test for me to
trunk that sets network_addr on the reciept of a reply to an invite on an
outbound call. and the 2 variables sip_reply_host and sip_reply_port




local and remote media port reflects the port being used between that leg
and it's remote connection eg the ip and port that the rtp stack was asked
to use.


On Wed, Dec 3, 2008 at 9:48 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:

>  I looked in the b-leg xml cdr and the ip address is not there (for
> signaling) it is only there
> for media (${remote_media_ip}) which is not the same thing now, is it?
>
> While we are at it, I noticed that the ${local_media_port} and
> ${remote_media_port}
> have the same value for each CDR (a or b leg). Shouldn't the first variable
> hold the port
> of the FS (on both legs) and the second variable the port of the client (in
> the a-leg) or the port of
> the provider (in the b-leg)?
>
> Anthony Minessale wrote:
>
> outgoing calls to not have an ip value set.
> if you want to store the dest ip in the cdr you need to set it as a custom
> variable and insert it
> into your template for csv cdr or it will just be there in xml cdr
>
> On Wed, Dec 3, 2008 at 8:18 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:
>
>> b-leg logging is enabled in the cdr module. but in the cdrs I cannot get
>> any variables that refer to the b-leg.
>>
>> I tried the second way using ${sip_to_host} and {bleg_sip_to_host} but :
>>
>> a) the variable returns the FS IP on the a-leg CDR (correctly)
>> b) the variable returns nothing on the b-leg CDR (empty). Shouldn't it
>> return the "to" host of the b-leg (my providers address)?
>>
>>
>> Anthony Minessale wrote:
>>
>>  2 options.
>> 1) enable b-leg logging on the cdr module.
>> 2) you can use the prefix bleg_ in a variable context to get to
>> caller_profile members
>> from the b leg.
>>
>> eg ${bleg_caller_id_name}
>>
>>
>> On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr <regs at kinetix.gr> wrote:
>>
>>> Hi,
>>>
>>>    I am making a simple bridge between two call legs :
>>>
>>> Client --(a-leg)--> FS --(b-leg)-->Provider
>>>
>>> How can I get information like network-address of the Provider,
>>> media-address,
>>> port used, media port used etc. from the second leg (b-leg)?
>>>
>>> Is all the information provided by the a-leg available for the b-leg as
>>> well? If, yese
>>> how can I access it? (and log it to my CDR file eventually)
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Dec  3 09:24:28 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 11:24:28 -0600
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<d0640d380812012224y57b30dddw1cd7aa0a522f2218@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
	<d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
Message-ID: <191c3a030812030924u457f934ep77bd70680f583fcd@mail.gmail.com>

please clean all the core.* files

reproduce the problem which will generate a core.xyz file (xyz is some
number)

run the command.

gdb /usr/local/freeswitch/bin/freeswitch core.xzy

when it loads issue the command

bt

and send me the output.








-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gab.tai at xtra.co.nz  Wed Dec  3 09:43:13 2008
From: gab.tai at xtra.co.nz (Gab Tai)
Date: Thu, 4 Dec 2008 06:43:13 +1300
Subject: [Freeswitch-users] Placing call to remote extension
Message-ID: <IJEDKKPJNGCNNCHBNDBHIEMNCBAA.gab.tai@xtra.co.nz>

Hi everyone,

I am Gab and just joined the group.  Also, I am new to FS but want to learn
and delve into the dept as fast as possible.  I have one last mile question
and was hoping I could pick from someone's wealth of knowledge and
understanding of the platform.

I have setup FS with 5 extensions as follows:


  1.. 1 extension [UA(a)] locally registered on the same network  NET(a) as
the realm of the FS(a)
  2.. 2 extensions [UA(b1) & UA(b2) ]  remotely registered to FS(a) from
subnet B
  3.. 2 extensions [UA(c1) & UA(c2) ] remotely registered to FS(a) from
subnet C
  4.. I am not using any provider
Current situation
  a.. All remote extensions can call UA(a) and transfer media (voice)
  b.. UA(a) cannot call remote extensions. Error message "Sofia cannot open
channel,; user not registered". But, please note that the user is actually
registered.
  c.. Remote extensions UA(b1) cannot call UA(b2) and cannot call UA(c1) nor
UA(c2)
  d.. Remote extensions UA(c1) cannot call UA(c2) and cannot call UA(b1) nor
UA(b2)

Need
  a.. How do I place call to remote extensions from local extensions?
  b.. How do I bridge media between 2 remote extensions, registered to FS(a)
from same network or different network?
I hope this is not too much for a starter but would greatly appreciate any
thoughts and/or guidance.

Sincere regards to all.
Gab
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From regs at kinetix.gr  Wed Dec  3 10:20:28 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 20:20:28 +0200
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>
References: <493689D8.9040708@kinetix.gr>	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>	<49369537.6040008@kinetix.gr>	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>	<4936AA33.7080301@kinetix.gr>
	<191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>
Message-ID: <4936CDEC.4010204@kinetix.gr>

I'll try the patch. Thank you for your time.

As for the local and remote media ports :

I have an endpoint with IP xxx.xxx.xxx.xxx and an FS box with IP 
yyy.yyy.yyy.yyy.
In a SIP bridge each side of the call leg between the two boxes will 
pick a udp port in order to send/receive traffic.

In my CDRs (a-leg) when I call the ${remote_media_port} and 
${local_media_port} it returns the same value (e.g. 18841) for both 
endpoints
(yyy.yyy.yyy.yyy and xxx.xxx.xxx.xxx).

In my b-leg CDR (let's say yyy.yyy.yyy.yyy to zzz.zzz.zzz.zzz) both 
variables hold the same value as well but a different
one than the a-leg's (e.g. 19871)

The way I thought it would happen is that each call leg would have a 
pair of different port numbers for the two variables
because :
 yyy would inform xxx that it should use port A
xxx would inform yyy that it should use port B
(that's one pair)
yyy would inform zzz that it should use port C
zzz would inform yyy that it should use port D
(that's another pair)

so for the a-leg : ${local_media_port} = A, ${remote_media_port} = B
for the b=leg : ${local_media_port} = C, ${remote_media_port} = D

Am I missing something?


Anthony Minessale wrote:
> It's not an unreasonabe request so i added a patch you can test for me 
> to trunk that sets network_addr on the reciept of a reply to an invite 
> on an outbound call. and the 2 variables sip_reply_host and sip_reply_port
>
>
>
>
> local and remote media port reflects the port being used between that 
> leg and it's remote connection eg the ip and port that the rtp stack 
> was asked to use.
>
>
> On Wed, Dec 3, 2008 at 9:48 AM, regs at kinetix.gr 
> <mailto:regs at kinetix.gr> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     I looked in the b-leg xml cdr and the ip address is not there (for
>     signaling) it is only there
>     for media (${remote_media_ip}) which is not the same thing now, is it?
>
>     While we are at it, I noticed that the ${local_media_port} and
>     ${remote_media_port}
>     have the same value for each CDR (a or b leg). Shouldn't the first
>     variable hold the port
>     of the FS (on both legs) and the second variable the port of the
>     client (in the a-leg) or the port of
>     the provider (in the b-leg)?
>
>     Anthony Minessale wrote:
>>     outgoing calls to not have an ip value set.
>>     if you want to store the dest ip in the cdr you need to set it as
>>     a custom variable and insert it
>>     into your template for csv cdr or it will just be there in xml cdr
>>
>>     On Wed, Dec 3, 2008 at 8:18 AM, regs at kinetix.gr
>>     <mailto:regs at kinetix.gr> <regs at kinetix.gr
>>     <mailto:regs at kinetix.gr>> wrote:
>>
>>         b-leg logging is enabled in the cdr module. but in the cdrs I
>>         cannot get any variables that refer to the b-leg.
>>
>>         I tried the second way using ${sip_to_host} and
>>         {bleg_sip_to_host} but :
>>
>>         a) the variable returns the FS IP on the a-leg CDR (correctly)
>>         b) the variable returns nothing on the b-leg CDR (empty).
>>         Shouldn't it return the "to" host of the b-leg (my providers
>>         address)?
>>
>>
>>         Anthony Minessale wrote:
>>>         2 options.
>>>         1) enable b-leg logging on the cdr module.
>>>         2) you can use the prefix bleg_ in a variable context to get
>>>         to caller_profile members
>>>         from the b leg.
>>>
>>>         eg ${bleg_caller_id_name}
>>>
>>>
>>>         On Wed, Dec 3, 2008 at 7:30 AM, regs at kinetix.gr
>>>         <mailto:regs at kinetix.gr> <regs at kinetix.gr
>>>         <mailto:regs at kinetix.gr>> wrote:
>>>
>>>             Hi,
>>>
>>>                I am making a simple bridge between two call legs :
>>>
>>>             Client --(a-leg)--> FS --(b-leg)-->Provider
>>>
>>>             How can I get information like network-address of the
>>>             Provider,
>>>             media-address,
>>>             port used, media port used etc. from the second leg (b-leg)?
>>>
>>>             Is all the information provided by the a-leg available
>>>             for the b-leg as
>>>             well? If, yese
>>>             how can I access it? (and log it to my CDR file eventually)
>>>
>>>             _______________________________________________
>>>             Freeswitch-users mailing list
>>>             Freeswitch-users at lists.freeswitch.org
>>>             <mailto:Freeswitch-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>         -- 
>>>         Anthony Minessale II
>>>
>>>         FreeSWITCH http://www.freeswitch.org/
>>>         ClueCon http://www.cluecon.com/
>>>
>>>         AIM: anthm
>>>         MSN:anthony_minessale at hotmail.com
>>>         <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>         GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>         <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>         IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>         FreeSWITCH Developer Conference
>>>         sip:888 at conference.freeswitch.org
>>>         <mailto:sip%3A888 at conference.freeswitch.org>
>>>         iax:guest at conference.freeswitch.org/888
>>>         <http://iax:guest at conference.freeswitch.org/888>
>>>         googletalk:conf+888 at conference.freeswitch.org
>>>         <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>         pstn:213-799-1400
>>>         ------------------------------------------------------------------------
>>>         _______________________________________________
>>>         Freeswitch-users mailing list
>>>         Freeswitch-users at lists.freeswitch.org
>>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         _______________________________________________
>>         Freeswitch-users mailing list
>>         Freeswitch-users at lists.freeswitch.org
>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Anthony Minessale II
>>
>>     FreeSWITCH http://www.freeswitch.org/
>>     ClueCon http://www.cluecon.com/
>>
>>     AIM: anthm
>>     MSN:anthony_minessale at hotmail.com
>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>>     FreeSWITCH Developer Conference
>>     sip:888 at conference.freeswitch.org
>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>     iax:guest at conference.freeswitch.org/888
>>     <http://iax:guest at conference.freeswitch.org/888>
>>     googletalk:conf+888 at conference.freeswitch.org
>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>     pstn:213-799-1400
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From lachezar.valchev at gmail.com  Wed Dec  3 10:27:54 2008
From: lachezar.valchev at gmail.com (Lachezar Valchev)
Date: Wed, 3 Dec 2008 20:27:54 +0200
Subject: [Freeswitch-users] CDR generated on maximum sessions reach
Message-ID: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>

Hello everybody,

I am new to the list and I hope I can find some help here, regarding an
issue I am experiencing with the CDRs written by Freeswitch.

The thing is, I am using the "max-sessions" and the "sessions-per-second"
parameters in switch.conf.xml to limit the maximum number of simultaneous
calls, I want to go through my Freeswitch server.

These options are working well, but I was expecting to have CDRs generated
for the calls, that are dropped when the limit is reached.
Unfortunately there is no such one.

My question is: Is there an option, which allows the generation of CDR, when
a call is dropped, because the maximum sessions limit is reached?

If there is no such option, is there any way to achieve it? Probably by
using the the mod_limit module? Can you, please tell me how to do it?

Any kind of advice is welcomed. Thank you in advance.

Regards,
Lachezar
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From msc at freeswitch.org  Wed Dec  3 11:02:29 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Dec 2008 11:02:29 -0800
Subject: [Freeswitch-users] CDR generated on maximum sessions reach
In-Reply-To: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>
References: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>
Message-ID: <87f2f3b90812031102i5e4f8f19q70de5eb7dfbf7959@mail.gmail.com>

On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev <
lachezar.valchev at gmail.com> wrote:

> Hello everybody,
>
> I am new to the list and I hope I can find some help here, regarding an
> issue I am experiencing with the CDRs written by Freeswitch.
>
> The thing is, I am using the "max-sessions" and the "sessions-per-second"
> parameters in switch.conf.xml to limit the maximum number of simultaneous
> calls, I want to go through my Freeswitch server.
>
> These options are working well, but I was expecting to have CDRs generated
> for the calls, that are dropped when the limit is reached.
> Unfortunately there is no such one.
>
> My question is: Is there an option, which allows the generation of CDR,
> when a call is dropped, because the maximum sessions limit is reached?
>
> If there is no such option, is there any way to achieve it? Probably by
> using the the mod_limit module? Can you, please tell me how to do it?
>

Lachezar,

These are good questions! I'll research them and let you know what I find
out.
-MC


>
> Any kind of advice is welcomed. Thank you in advance.
>
> Regards,
> Lachezar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec  3 11:41:04 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Dec 2008 11:41:04 -0800
Subject: [Freeswitch-users] CDR generated on maximum sessions reach
In-Reply-To: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>
References: <67a5ec7a0812031027s1a39c1a7y90dfe19e33bf193@mail.gmail.com>
Message-ID: <87f2f3b90812031141o6292a322j310f8dc0ff41d22e@mail.gmail.com>

On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev <
lachezar.valchev at gmail.com> wrote:

> Hello everybody,
>
> I am new to the list and I hope I can find some help here, regarding an
> issue I am experiencing with the CDRs written by Freeswitch.
>
> The thing is, I am using the "max-sessions" and the "sessions-per-second"
> parameters in switch.conf.xml to limit the maximum number of simultaneous
> calls, I want to go through my Freeswitch server.
>
> These options are working well, but I was expecting to have CDRs generated
> for the calls, that are dropped when the limit is reached.
> Unfortunately there is no such one.
>
> My question is: Is there an option, which allows the generation of CDR,
> when a call is dropped, because the maximum sessions limit is reached?


This is not possible. A CDR cannot be generated without a session, and a
session will not be generated if the max sessions limit has already been
reached...


>
>
> If there is no such option, is there any way to achieve it? Probably by
> using the the mod_limit module? Can you, please tell me how to do it?
>

mod_limit is most definitely your best option at this point. If you haven't
read this yet please do:
http://wiki.freeswitch.org/wiki/Mod_limit

In the example on that page, you have a "limit_exceeded" extension which
would show up in your CDR, or you can set a specific channel variable which
will magically show up in an XML CDR. (You can modify CSV CDRs to have any
custom channel variables as well. See
http://wiki.freeswitch.org/wiki/Cdrwhich refers to the <templates>
section of
conf/autoload_configs/cdr_csv.conf.xml)

Try adding the limit app in your dialplan and have the limit_exceeded
extension as well. You could set the limit really low for the sake of
testing before setting it to the value necessary for your production
deployment.

Let us know how it goes.

-MC


>
> Any kind of advice is welcomed. Thank you in advance.
>
> Regards,
> Lachezar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at suspicious.org  Wed Dec  3 12:01:47 2008
From: krice at suspicious.org (Ken Rice)
Date: Wed, 03 Dec 2008 14:01:47 -0600
Subject: [Freeswitch-users] CDR generated on maximum sessions reach
In-Reply-To: <87f2f3b90812031141o6292a322j310f8dc0ff41d22e@mail.gmail.com>
Message-ID: <C55C41CB.4ACEA%krice@suspicious.org>

From: Michael Collins <msc at freeswitch.org>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Wed, 3 Dec 2008 11:41:04 -0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] CDR generated on maximum sessions reach



On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev
<lachezar.valchev at gmail.com> wrote:
> Hello everybody,
> 
> I am new to the list and I hope I can find some help here, regarding an issue
> I am experiencing with the CDRs written by Freeswitch.
> 
> The thing is, I am using the "max-sessions" and the "sessions-per-second"
> parameters in switch.conf.xml to limit the maximum number of simultaneous
> calls, I want to go through my Freeswitch server.
> 
> These options are working well, but I was expecting to have CDRs generated for
> the calls, that are dropped when the limit is reached.
> Unfortunately there is no such one.
> 
> My question is: Is there an option, which allows the generation of CDR, when a
> call is dropped, because the maximum sessions limit is reached?

This is not possible. A CDR cannot be generated without a session, and a
session will not be generated if the max sessions limit has already been
reached...
 
> 
> 
> If there is no such option, is there any way to achieve it? Probably by using
> the the mod_limit module? Can you, please tell me how to do it?

mod_limit is most definitely your best option at this point. If you haven't
read this yet please do:
http://wiki.freeswitch.org/wiki/Mod_limit

In the example on that page, you have a "limit_exceeded" extension which
would show up in your CDR, or you can set a specific channel variable which
will magically show up in an XML CDR. (You can modify CSV CDRs to have any
custom channel variables as well. See http://wiki.freeswitch.org/wiki/Cdr
which refers to the <templates> section of
conf/autoload_configs/cdr_csv.conf.xml)

Try adding the limit app in your dialplan and have the limit_exceeded
extension as well. You could set the limit really low for the sake of
testing before setting it to the value necessary for your production
deployment.

------SNIP-------

Michael?s advise here is dead on... The whole point of the session limiter
(and SPS limiters) is to keep the box from melting down... Say in the event
of a DoS (or a telemarketer)... If you want to do a soft limit that?s just
want mod_limit is for and will allow you to do 2 things when used in
conjunction with the max_sessions and sps limits... 1) Soft Limit and still
log the calls 2) still have a hard limit that keeps the box for falling over
dead... 

You can do this with a simple step at the top of your dialplan that
mod_limits everything together...

K
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From anthony.minessale at gmail.com  Wed Dec  3 12:11:40 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 14:11:40 -0600
Subject: [Freeswitch-users] How to get info from the b-leg
In-Reply-To: <4936CDEC.4010204@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
	<49369537.6040008@kinetix.gr>
	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
	<4936AA33.7080301@kinetix.gr>
	<191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>
	<4936CDEC.4010204@kinetix.gr>
Message-ID: <191c3a030812031211q41b69501q5b1b601adf9d0b4d@mail.gmail.com>

looks like a typo in the code.  I guess nobody ever looked at that field
before.
it should be fixed in r10582



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Wed Dec  3 12:33:25 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Dec 2008 12:33:25 -0800
Subject: [Freeswitch-users] Placing call to remote extension
In-Reply-To: <IJEDKKPJNGCNNCHBNDBHIEMNCBAA.gab.tai@xtra.co.nz>
References: <IJEDKKPJNGCNNCHBNDBHIEMNCBAA.gab.tai@xtra.co.nz>
Message-ID: <87f2f3b90812031233l6cb00525geea2a3ede4c045bd@mail.gmail.com>

Hi Gab!

Welcome to FreeSWITCH. Thanks for your questions. I'm trying to learn all of
this stuff and help others so bear with me while I research these and help
you find the answers.

BTW, are you on IRC? you can visit us for realtime help, #freeswitch on
irc.freenode.net
-MC (mercutioviz on irc)

On Wed, Dec 3, 2008 at 9:43 AM, Gab Tai <gab.tai at xtra.co.nz> wrote:

>  Hi everyone,
>
> I am Gab and just joined the group.  Also, I am new to FS but want to learn
> and delve into the dept as fast as possible.  I have one last mile question
> and was hoping I could pick from someone's wealth of knowledge and
> understanding of the platform.
>
> I have setup FS with 5 extensions as follows:
>
>    1. 1 extension [UA(a)] locally registered on the same network  NET(a)
>    as the realm of the FS(a)
>    2. 2 extensions [UA(b1) & UA(b2) ]  remotely registered to FS(a) from
>    subnet B
>    3. 2 extensions [UA(c1) & UA(c2) ] remotely registered to FS(a) from
>    subnet C
>    4. I am not using any provider
>
> *Current situation*
>
>    - All remote extensions can call UA(a) and transfer media (voice)
>    - UA(a) cannot call remote extensions. Error message "Sofia cannot open
>    channel,; user not registered". But, please note that the user is actually
>    registered.
>    - Remote extensions UA(b1) cannot call UA(b2) and cannot call UA(c1)
>    nor UA(c2)
>    - Remote extensions UA(c1) cannot call UA(c2) and cannot call UA(b1)
>    nor UA(b2)
>
>
> *Need*
>
>    - How do I place call to remote extensions from local extensions?
>    - How do I bridge media between 2 remote extensions, registered to
>    FS(a) from same network or different network?
>
> I hope this is not too much for a starter but would greatly appreciate any
> thoughts and/or guidance.
>
> Sincere regards to all.
> Gab
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jkr888 at gmail.com  Wed Dec  3 12:48:04 2008
From: jkr888 at gmail.com (Johny Kadarisman)
Date: Wed, 3 Dec 2008 15:48:04 -0500
Subject: [Freeswitch-users] Redirect calls between FS
In-Reply-To: <b081cc8f0811290318h5b27af05h531cd80230dd04d2@mail.gmail.com>
References: <b081cc8f0811290318h5b27af05h531cd80230dd04d2@mail.gmail.com>
Message-ID: <b0a271780812031248x4fd7c240rd97ad16bd6ecb34b@mail.gmail.com>

Is these apps will work for you?

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect


On Sat, Nov 29, 2008 at 6:18 AM, Barray McKee <barraymckee at gmail.com> wrote:
> Hello,
>
> I am implementing 2 load balancing FS behind a pair of sip proxies.  Since
> traffic is routed to one of the two FS on a round-robin basis, I need a
> mechanism to reroute call from one FS to another FS under a specific special
> circumstance.  I am think to use the bridge command with bypass_media =
> true, so that the call can just pass through one FS and reach the second
> FS.   Is this the approach that I should use?  I am wondering if there is a
> better approach out there that I just can't think of.
>
> Thanks in advance,
> Barray
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From regs at kinetix.gr  Wed Dec  3 13:24:18 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Wed, 03 Dec 2008 23:24:18 +0200
Subject: [Freeswitch-users] How to get info from the b-leg [PATCHED -
 FIXED]
In-Reply-To: <191c3a030812031211q41b69501q5b1b601adf9d0b4d@mail.gmail.com>
References: <493689D8.9040708@kinetix.gr>	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>	<49369537.6040008@kinetix.gr>	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>	<4936AA33.7080301@kinetix.gr>	<191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>	<4936CDEC.4010204@kinetix.gr>
	<191c3a030812031211q41b69501q5b1b601adf9d0b4d@mail.gmail.com>
Message-ID: <4936F902.1070905@kinetix.gr>

I tested both patches from the trunk : network_addr is set to the remote 
IP on the b-leg
and local media port and remote media port hold the correct values when 
called.
Both pathces work like a charm!
Thanks for your time and effort :)

Anthony Minessale wrote:
>
>
> looks like a typo in the code.  I guess nobody ever looked at that 
> field before.
> it should be fixed in r10582
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From msc at freeswitch.org  Wed Dec  3 13:32:25 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Dec 2008 13:32:25 -0800
Subject: [Freeswitch-users] How to get info from the b-leg [PATCHED -
	FIXED]
In-Reply-To: <4936F902.1070905@kinetix.gr>
References: <493689D8.9040708@kinetix.gr>
	<191c3a030812030553u3a166434g4c51a5c938a2c315@mail.gmail.com>
	<49369537.6040008@kinetix.gr>
	<191c3a030812030737od8e0311q63c6944cc411dcc1@mail.gmail.com>
	<4936AA33.7080301@kinetix.gr>
	<191c3a030812030913ka04c04dxaada63e5b8f84fe4@mail.gmail.com>
	<4936CDEC.4010204@kinetix.gr>
	<191c3a030812031211q41b69501q5b1b601adf9d0b4d@mail.gmail.com>
	<4936F902.1070905@kinetix.gr>
Message-ID: <87f2f3b90812031332t291a96d5heafbfc2914e48f09@mail.gmail.com>

And thank you for testing and being gracious! :)
-MC

On Wed, Dec 3, 2008 at 1:24 PM, regs at kinetix.gr <regs at kinetix.gr> wrote:

>  I tested both patches from the trunk : network_addr is set to the remote
> IP on the b-leg
> and local media port and remote media port hold the correct values when
> called.
> Both pathces work like a charm!
> Thanks for your time and effort :)
>
> Anthony Minessale wrote:
>
>
>
> looks like a typo in the code.  I guess nobody ever looked at that field
> before.
> it should be fixed in r10582
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kkielhofner at star2star.com  Wed Dec  3 14:26:15 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Wed, 3 Dec 2008 17:26:15 -0500
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <87f2f3b90812021718j5aae69aav6dd8ee7953e2b1ff@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
	<87f2f3b90812021718j5aae69aav6dd8ee7953e2b1ff@mail.gmail.com>
Message-ID: <2d9149cd0812031426h71ccaddqfbd6d6af34e8eaa7@mail.gmail.com>

On Tue, Dec 2, 2008 at 8:18 PM, Michael Collins <msc at freeswitch.org> wrote:
> Kristian,
>
> Are you on the IRC channel by any chance?
> -MC (IRC: mercutioviz)
>

Me? Never!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From sobolewski at gmail.com  Wed Dec  3 14:33:24 2008
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Wed, 3 Dec 2008 23:33:24 +0100
Subject: [Freeswitch-users] screen_bit is always true
Message-ID: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>

Hi

I was trying  to create extension in which I would check whether
privacy=full in Remote-Party-ID header is set.
So I made this.

   <condition field="${screen_bit}" expression="^true$" break="never">
          <action application="set"
data="origination_caller_id_number=Anonymous"/>
   </condition>

But screen_bit is always true, regardless RPID privacy value.

        mod_dialplan_xml.c:117 parse_exten() Regex: [tooser]
${screen_bit}(true) =~ /^true$/

As it wasn't strange enough, in cdr_csv screen_bit take "true" or
"false" (as it should).

Is it a bug or me doing something wrong?

--
regards
Piotr Sobolewski



From krice at suspicious.org  Wed Dec  3 16:14:57 2008
From: krice at suspicious.org (Ken Rice)
Date: Wed, 03 Dec 2008 18:14:57 -0600
Subject: [Freeswitch-users] screen_bit is always true
In-Reply-To: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
Message-ID: <C55C7D21.4AD00%krice@suspicious.org>

The screen bit is a trust bit... ie: do we trust the RPID we got from the
upstream or not

K


> From: Piotr Sobolewski <sobolewski at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 3 Dec 2008 23:33:24 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] screen_bit is always true
> 
> Hi
> 
> I was trying  to create extension in which I would check whether
> privacy=full in Remote-Party-ID header is set.
> So I made this.
> 
>    <condition field="${screen_bit}" expression="^true$" break="never">
>           <action application="set"
> data="origination_caller_id_number=Anonymous"/>
>    </condition>
> 
> But screen_bit is always true, regardless RPID privacy value.
> 
>         mod_dialplan_xml.c:117 parse_exten() Regex: [tooser]
> ${screen_bit}(true) =~ /^true$/
> 
> As it wasn't strange enough, in cdr_csv screen_bit take "true" or
> "false" (as it should).
> 
> Is it a bug or me doing something wrong?
> 
> --
> regards
> Piotr Sobolewski
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Wed Dec  3 17:38:13 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Dec 2008 19:38:13 -0600
Subject: [Freeswitch-users] Fax and Freeswitch: What is the status,
	what works?
In-Reply-To: <2d9149cd0812031426h71ccaddqfbd6d6af34e8eaa7@mail.gmail.com>
References: <5e414ed0812020940l34520124p8c4b9039cfdedfaf@mail.gmail.com>
	<87f2f3b90812021232j23db1635s9cd9fa8c1c4b23e2@mail.gmail.com>
	<2d9149cd0812021328pbf6dbe4s8d703f77c9ba71c3@mail.gmail.com>
	<87f2f3b90812021718j5aae69aav6dd8ee7953e2b1ff@mail.gmail.com>
	<2d9149cd0812031426h71ccaddqfbd6d6af34e8eaa7@mail.gmail.com>
Message-ID: <BC166D25-7051-4E3F-B27B-DE55984A1E63@freeswitch.org>

You scared? muhahaha  Will I see you at ClueCon 09, its the first week  
in Aug. again.... in Chicago.

/b

On Dec 3, 2008, at 4:26 PM, Kristian Kielhofner wrote:

> Me? Never!




From brian at freeswitch.org  Wed Dec  3 17:38:53 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Dec 2008 19:38:53 -0600
Subject: [Freeswitch-users] screen_bit is always true
In-Reply-To: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
References: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
Message-ID: <A914D032-DC2F-4DF8-9808-4DE6EBF9A08E@freeswitch.org>

I think we covered this on IRC already didn't we?

/b

On Dec 3, 2008, at 4:33 PM, Piotr Sobolewski wrote:

>
> Is it a bug or me doing something wrong?




From sobolewski at gmail.com  Wed Dec  3 18:02:40 2008
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Thu, 4 Dec 2008 03:02:40 +0100
Subject: [Freeswitch-users] screen_bit is always true
In-Reply-To: <C55C7D21.4AD00%krice@suspicious.org>
References: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
	<C55C7D21.4AD00%krice@suspicious.org>
Message-ID: <3666ca0d0812031802o754ed22eub5435345e542153e@mail.gmail.com>

On Thu, Dec 4, 2008 at 1:14 AM, Ken Rice <krice at suspicious.org> wrote:
> The screen bit is a trust bit... ie: do we trust the RPID we got from the
> upstream or not
>
> K

I had privacy_hide_number in cdr_csv and I was thinking it was
screen_bit, all that confused me.
Now I understand where I was wrong.

BTW: is there a way to remove RPID header?


-- 
Piotr Sobolewski
sobolewski at gmail.com



From brian at freeswitch.org  Wed Dec  3 18:12:27 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Dec 2008 20:12:27 -0600
Subject: [Freeswitch-users] screen_bit is always true
In-Reply-To: <3666ca0d0812031802o754ed22eub5435345e542153e@mail.gmail.com>
References: <3666ca0d0812031433l1811dc83t606e085a58763ee@mail.gmail.com>
	<C55C7D21.4AD00%krice@suspicious.org>
	<3666ca0d0812031802o754ed22eub5435345e542153e@mail.gmail.com>
Message-ID: <4F994CD4-43CF-4AC9-8EC2-A66AD715A1B0@freeswitch.org>

I already told you this one on IRC too :P  email is too slow today :)

/b

On Dec 3, 2008, at 8:02 PM, Piotr Sobolewski wrote:

> BTW: is there a way to remove RPID header?




From ack at telefonica.net  Wed Dec  3 17:03:08 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Thu, 04 Dec 2008 02:03:08 +0100
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
Message-ID: <1228352588.25709.42.camel@develop4>

Hi guys, 
 
 I've a strange issue with FS , version svn -r10584 ,
when FS bridges a call first 3 seconds of audio are missing , looks that
only happens on PSTN calls and using ringback or transfer_ringback. This
only happens in calls from PSTN , not from VOIP. Some scenarios i tried
to isolate this issue :


- Issue

PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN

- Good setting bypass_media before run bridge but i need rtp in FS path

PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN

- Good 

PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback -> PSTN

- Always good

VOIP --> FS ( brigde ) -> PSTN


Dialplan has nothing wrong ( i guess ):

<extension name="Transfers">
    <condition field="destination_number" expression="^1??XXXXXXXXXX$">
      <action application="answer"/>
      <action application="speak" data="cepstral|Allison-8kHz|blah"/>
      <action application="set" data="hangup_after_bridge=false"/>
      <action application="set" data="playback_terminators=#"/>
      <action application="set" data="ringback=$${us-ring}"/>
      <action application="set" data="transfer_ringback=
$${hold_music}"/>
      <action application="set" data="effective_caller_id_name=
${caller_id_name}"/>
      <action application="set" data="effective_caller_id_number=
${caller_id_number}"/>
      <action application="set" data="originate_timeout=30"/>
      <action application="set" data="call_timeout=30"/>
      <action application="bridge"
data="sofia/default/18008226235 at PSTN_GW"/>
      <action application="speak" data="cepstral|Allison-8kHz|Transfer
finished"/>
      <action application="hangup"/>
    </condition>
  </extension>



Any ideas ?

Attached log of FS ( F8 from console ).


Thanks in advance !

-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

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From anthony.minessale at gmail.com  Wed Dec  3 20:12:00 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Dec 2008 22:12:00 -0600
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <1228352588.25709.42.camel@develop4>
References: <1228352588.25709.42.camel@develop4>
Message-ID: <191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>

what does PSTN represent?

I know what the PSTN is but how are you reaching it?
is it TDM, SIP etc... what gateway type other details.


On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <ack at telefonica.net> wrote:

> Hi guys,
>
>  I've a strange issue with FS , version svn -r10584 ,
> when FS bridges a call first 3 seconds of audio are missing , looks that
> only happens on PSTN calls and using ringback or transfer_ringback. This
> only happens in calls from PSTN , not from VOIP. Some scenarios i tried
> to isolate this issue :
>
>
> - Issue
>
> PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>
> - Good setting bypass_media before run bridge but i need rtp in FS path
>
> PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>
> - Good
>
> PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback -> PSTN
>
> - Always good
>
> VOIP --> FS ( brigde ) -> PSTN
>
>
> Dialplan has nothing wrong ( i guess ):
>
> <extension name="Transfers">
>    <condition field="destination_number" expression="^1??XXXXXXXXXX$">
>      <action application="answer"/>
>      <action application="speak" data="cepstral|Allison-8kHz|blah"/>
>      <action application="set" data="hangup_after_bridge=false"/>
>      <action application="set" data="playback_terminators=#"/>
>      <action application="set" data="ringback=$${us-ring}"/>
>      <action application="set" data="transfer_ringback=
> $${hold_music}"/>
>      <action application="set" data="effective_caller_id_name=
> ${caller_id_name}"/>
>      <action application="set" data="effective_caller_id_number=
> ${caller_id_number}"/>
>      <action application="set" data="originate_timeout=30"/>
>      <action application="set" data="call_timeout=30"/>
>      <action application="bridge"
> data="sofia/default/18008226235 at PSTN_GW"/>
>      <action application="speak" data="cepstral|Allison-8kHz|Transfer
> finished"/>
>      <action application="hangup"/>
>    </condition>
>  </extension>
>
>
>
> Any ideas ?
>
> Attached log of FS ( F8 from console ).
>
>
> Thanks in advance !
>
> --
> Angel Carpintero
> ack ( at ) telefonica ( dot ) net
>
> Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ack at telefonica.net  Wed Dec  3 20:46:21 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Thu, 04 Dec 2008 05:46:21 +0100
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
References: <1228352588.25709.42.camel@develop4>
	<191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
Message-ID: <1228365981.25709.60.camel@develop4>

No TDM , all is SIP :


PSTN ---> Sip Proxy_A --> FS ( brigde ) ringback/transfer_ringback 
-> Sip Proxy_B --> PSTN 


In logfile i think you can get some details about Media Gateways
( Sonus ) PSTN inbound / outbound is provided by Level3. 

I can get a capture of a call if you want, in capture the audio is not 
missing, issue with :

- rtp buffer ?
- Sonus ? 

Let me know anything you need so i can provide a log or create a new
scenario.


Thanks,

El mi?, 03-12-2008 a las 22:12 -0600, Anthony Minessale escribi?:
> what does PSTN represent? 
> 
> I know what the PSTN is but how are you reaching it?
> is it TDM, SIP etc... what gateway type other details.
> 
> 
> On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <ack at telefonica.net>
> wrote:
>         Hi guys,
>         
>          I've a strange issue with FS , version svn -r10584 ,
>         when FS bridges a call first 3 seconds of audio are missing ,
>         looks that
>         only happens on PSTN calls and using ringback or
>         transfer_ringback. This
>         only happens in calls from PSTN , not from VOIP. Some
>         scenarios i tried
>         to isolate this issue :
>         
>         
>         - Issue
>         
>         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>         
>         - Good setting bypass_media before run bridge but i need rtp
>         in FS path
>         
>         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>         
>         - Good
>         
>         PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback ->
>         PSTN
>         
>         - Always good
>         
>         VOIP --> FS ( brigde ) -> PSTN
>         
>         
>         Dialplan has nothing wrong ( i guess ):
>         
>         <extension name="Transfers">
>            <condition field="destination_number"
>         expression="^1??XXXXXXXXXX$">
>              <action application="answer"/>
>              <action application="speak" data="cepstral|Allison-8kHz|
>         blah"/>
>              <action application="set"
>         data="hangup_after_bridge=false"/>
>              <action application="set" data="playback_terminators=#"/>
>              <action application="set" data="ringback=$${us-ring}"/>
>              <action application="set" data="transfer_ringback=
>         $${hold_music}"/>
>              <action application="set" data="effective_caller_id_name=
>         ${caller_id_name}"/>
>              <action application="set"
>         data="effective_caller_id_number=
>         ${caller_id_number}"/>
>              <action application="set" data="originate_timeout=30"/>
>              <action application="set" data="call_timeout=30"/>
>              <action application="bridge"
>         data="sofia/default/18008226235 at PSTN_GW"/>
>              <action application="speak" data="cepstral|Allison-8kHz|
>         Transfer
>         finished"/>
>              <action application="hangup"/>
>            </condition>
>          </extension>
>         
>         
>         
>         Any ideas ?
>         
>         Attached log of FS ( F8 from console ).
>         
>         
>         Thanks in advance !
>         
>         --
>         Angel Carpintero
>         ack ( at ) telefonica ( dot ) net
>         
>         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
>         6EF1 B90D
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400

-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

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From pieter_eduard at biznetnetworks.com  Thu Dec  4 00:03:36 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Thu, 04 Dec 2008 15:03:36 +0700
Subject: [Freeswitch-users] freeswitch enum
Message-ID: <49378ED8.9010404@biznetnetworks.com>

Hi,

I'm trying to query freeswitch  to use my bind base enum server but i'm 
having trouble to query the enum from Freeswitch CLI.

this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after 
restart the dns, i have this :

[root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
Using domain server:
Name: localhost
Address: 127.0.0.1#53
Aliases:

8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel" 
"!^.*$!tel:\\12345678!" .
8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip" 
"!^.*$!sip:1000 at test.com!" .

and if use the ip, the result is the same :
[root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
Using domain server:
Name: 1.2.3.4
Address: 1.2.3.4#53
Aliases:

8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip" 
"!^.*$!sip:1000 at test.com!" .
8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel" 
"!^.*$!tel:\\12345678!" .

But if i try to query from the freeswitch CLI that's installed in the 
same box, i get this :

freeswitch at fsbox> enum 12345678 localhost
API CALL [enum(12345678 localhost)] output:
No Match!

freeswitch at fsbox> enum 12345678 1.2.3.4
API CALL [enum(12345678 1.2.3.4)] output:
No Match!

even if i change the enum.conf.xml dns root with my enum ip and reload 
the freeswitch i still get the same error.

Can anyone help me on this? i just want the fs to query on my local enum 
or to query on different enum server.

Regards,
-Pieter-




From krice at suspicious.org  Thu Dec  4 00:13:08 2008
From: krice at suspicious.org (Ken Rice)
Date: Thu, 04 Dec 2008 02:13:08 -0600
Subject: [Freeswitch-users] freeswitch enum
In-Reply-To: <49378ED8.9010404@biznetnetworks.com>
Message-ID: <C55CED34.4AD19%krice@suspicious.org>

Ok 1) Overriding the e164.arpa is probably not a good Idea... 2) the second
param for the enum command is a domain not a Server IP address

Example: enum 1234567890 e164.org

You should probably set up your enum records on your own private domain or
use a real domain that you own... If you have to use something like e164.int
as the domain then tell your name server its the SOA for that domain and set
all your records up there...

Then in FreeSwitch tell it the default domain is e164.int

Chances are if you do something like enum 18005551212 e164.org it will work
correctly...


> From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 04 Dec 2008 15:03:36 +0700
> To: "freeswitch-users at lists.freeswitch.org"
> <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] freeswitch enum
> 
> Hi,
> 
> I'm trying to query freeswitch  to use my bind base enum server but i'm
> having trouble to query the enum from Freeswitch CLI.
> 
> this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after
> restart the dns, i have this :
> 
> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
> Using domain server:
> Name: localhost
> Address: 127.0.0.1#53
> Aliases:
> 
> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
> "!^.*$!tel:\\12345678!" .
> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
> "!^.*$!sip:1000 at test.com!" .
> 
> and if use the ip, the result is the same :
> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
> Using domain server:
> Name: 1.2.3.4
> Address: 1.2.3.4#53
> Aliases:
> 
> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
> "!^.*$!sip:1000 at test.com!" .
> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
> "!^.*$!tel:\\12345678!" .
> 
> But if i try to query from the freeswitch CLI that's installed in the
> same box, i get this :
> 
> freeswitch at fsbox> enum 12345678 localhost
> API CALL [enum(12345678 localhost)] output:
> No Match!
> 
> freeswitch at fsbox> enum 12345678 1.2.3.4
> API CALL [enum(12345678 1.2.3.4)] output:
> No Match!
> 
> even if i change the enum.conf.xml dns root with my enum ip and reload
> the freeswitch i still get the same error.
> 
> Can anyone help me on this? i just want the fs to query on my local enum
> or to query on different enum server.
> 
> Regards,
> -Pieter-
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From carole.olivier at enst.fr  Thu Dec  4 00:28:04 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 4 Dec 2008 00:28:04 -0800 (PST)
Subject: [Freeswitch-users] re gistration and calls through different user
 agent
Message-ID: <20829148.post@talk.nabble.com>


Hello,

I have recently installed freeswitch on Opensuse 10.3.

I have a question about the sofia agents which are already present in the
default installation:
I have 2 snom phones (each with a user1001 and user1002), I configured one
such that it registered on the default port 5060 (so on the user agent
internal) and the other one on the port 5080 (on the external user agent).
The aim was to see what happen.

Since both were able to register I guess that the directory
/directory/default/ is used by both sofia agent else user1002 should not be
able to register, isn't? I am confused here because I would have said that
the directory /directory/default/ was just for the users that register on
internal...

I had a second problem: if user1001 tries to call user1002 it did not work
(but the opposite is ok). Has this something to do with the dialplan? 

Thanks,
Carole
-- 
View this message in context: http://www.nabble.com/registration-and-calls-through-different-user-agent-tp20829148p20829148.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From pieter_eduard at biznetnetworks.com  Thu Dec  4 01:12:36 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Thu, 04 Dec 2008 16:12:36 +0700
Subject: [Freeswitch-users] freeswitch enum
In-Reply-To: <C55CED34.4AD19%krice@suspicious.org>
References: <C55CED34.4AD19%krice@suspicious.org>
Message-ID: <49379F04.10704@biznetnetworks.com>

Ken,

I have done what you suggested bellow, for security reason i can not 
paste the query using the domain, just the localhost and i change the 
public  ip to 1.2.3.4.
So i already setup a real name server with  real domain and it uses  
public ip, configure the 7.6.5.4.3.2.1.e164.arpa as a domain with my 
name server as  SOA and NS record and yet the freeswitch CLI still give 
me no match when i try to query it using my name server.

And yes, if i query 18005551212 e164.org it works like a charm ;-)

Any other suggestions?

Regards,

-Pieter-
I
Ken Rice wrote:
> Ok 1) Overriding the e164.arpa is probably not a good Idea... 2) the second
> param for the enum command is a domain not a Server IP address
>
> Example: enum 1234567890 e164.org
>
> You should probably set up your enum records on your own private domain or
> use a real domain that you own... If you have to use something like e164.int
> as the domain then tell your name server its the SOA for that domain and set
> all your records up there...
>
> Then in FreeSwitch tell it the default domain is e164.int
>
> Chances are if you do something like enum 18005551212 e164.org it will work
> correctly...
>
>
>   
>> From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Thu, 04 Dec 2008 15:03:36 +0700
>> To: "freeswitch-users at lists.freeswitch.org"
>> <freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] freeswitch enum
>>
>> Hi,
>>
>> I'm trying to query freeswitch  to use my bind base enum server but i'm
>> having trouble to query the enum from Freeswitch CLI.
>>
>> this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after
>> restart the dns, i have this :
>>
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
>> Using domain server:
>> Name: localhost
>> Address: 127.0.0.1#53
>> Aliases:
>>
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>>
>> and if use the ip, the result is the same :
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
>> Using domain server:
>> Name: 1.2.3.4
>> Address: 1.2.3.4#53
>> Aliases:
>>
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>>
>> But if i try to query from the freeswitch CLI that's installed in the
>> same box, i get this :
>>
>> freeswitch at fsbox> enum 12345678 localhost
>> API CALL [enum(12345678 localhost)] output:
>> No Match!
>>
>> freeswitch at fsbox> enum 12345678 1.2.3.4
>> API CALL [enum(12345678 1.2.3.4)] output:
>> No Match!
>>
>> even if i change the enum.conf.xml dns root with my enum ip and reload
>> the freeswitch i still get the same error.
>>
>> Can anyone help me on this? i just want the fs to query on my local enum
>> or to query on different enum server.
>>
>> Regards,
>> -Pieter-
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> .
>
>   
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From krice at suspicious.org  Thu Dec  4 01:26:33 2008
From: krice at suspicious.org (Ken Rice)
Date: Thu, 04 Dec 2008 03:26:33 -0600
Subject: [Freeswitch-users] freeswitch enum
In-Reply-To: <49379F04.10704@biznetnetworks.com>
Message-ID: <C55CFE69.4AD1F%krice@suspicious.org>

No other suggestion your DNS setup is broken... You have proven that
freeswitch is working by querying e164.org...

Again you need to use a domain other than one that ends in .arpa that is a
reservered TLD and you will break things... Use the .int TLD that?s what its
for and if you are really worried about leaking things to the outside world
put an ACL on your DNS server to not let anyone outside your netblocks query
it... 2) resolv.conf on your fs box must point only at your name servers
that are SOA (either primary or secondary) for your private domain and much
not contain servers of your ISP that will cause lookup failures...
3) and once again you do not specify the IP address you are querying using
the enum command... You specify the number and the root (root being the
domain ie: mydomain.int)

Beyond that you need to check out other resources appropriate for setting up
enum records for your specific DNS implementation



From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 04 Dec 2008 16:12:36 +0700
To: "freeswitch-users at lists.freeswitch.org"
<freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch enum

Ken,

I have done what you suggested bellow, for security reason i can not paste
the query using the domain, just the localhost and i change the public  ip
to 1.2.3.4.
So i already setup a real name server with  real domain and it uses public
ip, configure the 7.6.5.4.3.2.1.e164.arpa as a domain with my name server as
SOA and NS record and yet the freeswitch CLI still give me no match when i
try to query it using my name server.

And yes, if i query 18005551212 e164.org it works like a charm ;-)

Any other suggestions?

Regards,

-Pieter-
I
Ken Rice wrote: 
>  
> Ok 1) Overriding the e164.arpa is probably not a good Idea... 2) the second
> param for the enum command is a domain not a Server IP address
> 
> Example: enum 1234567890 e164.org
> 
> You should probably set up your enum records on your own private domain or
> use a real domain that you own... If you have to use something like e164.int
> as the domain then tell your name server its the SOA for that domain and set
> all your records up there...
> 
> Then in FreeSwitch tell it the default domain is e164.int
> 
> Chances are if you do something like enum 18005551212 e164.org it will work
> correctly...
> 
> 
>   
>  
>>  
>> From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
>> <mailto:pieter_eduard at biznetnetworks.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> Date: Thu, 04 Dec 2008 15:03:36 +0700
>> To: "freeswitch-users at lists.freeswitch.org"
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] freeswitch enum
>> 
>> Hi,
>> 
>> I'm trying to query freeswitch  to use my bind base enum server but i'm
>> having trouble to query the enum from Freeswitch CLI.
>> 
>> this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after
>> restart the dns, i have this :
>> 
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
>> Using domain server:
>> Name: localhost
>> Address: 127.0.0.1#53
>> Aliases:
>> 
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 
>> and if use the ip, the result is the same :
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
>> Using domain server:
>> Name: 1.2.3.4
>> Address: 1.2.3.4#53
>> Aliases:
>> 
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 
>> But if i try to query from the freeswitch CLI that's installed in the
>> same box, i get this :
>> 
>> freeswitch at fsbox> enum 12345678 localhost
>> API CALL [enum(12345678 localhost)] output:
>> No Match!
>> 
>> freeswitch at fsbox> enum 12345678 1.2.3.4
>> API CALL [enum(12345678 1.2.3.4)] output:
>> No Match!
>> 
>> even if i change the enum.conf.xml dns root with my enum ip and reload
>> the freeswitch i still get the same error.
>> 
>> Can anyone help me on this? i just want the fs to query on my local enum
>> or to query on different enum server.
>> 
>> Regards,
>> -Pieter-
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>>  
>>  
>> 
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> .
>> 
>>   
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

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From krice at suspicious.org  Thu Dec  4 01:51:05 2008
From: krice at suspicious.org (Ken Rice)
Date: Thu, 04 Dec 2008 03:51:05 -0600
Subject: [Freeswitch-users] freeswitch enum
In-Reply-To: <C55CFE69.4AD1F%krice@suspicious.org>
Message-ID: <C55D0429.4AD21%krice@suspicious.org>

Ooops lemme correct myself...  .int is for international orgs and you
shouldn?t use that either... Try .localnet or .rofl something that is not
and probably never will be allocated.....

K



From: Ken Rice <krice at suspicious.org>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 04 Dec 2008 03:26:33 -0600
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch enum

No other suggestion your DNS setup is broken... You have proven that
freeswitch is working by querying e164.org...

Again you need to use a domain other than one that ends in .arpa that is a
reservered TLD and you will break things... Use the .int TLD that?s what its
for and if you are really worried about leaking things to the outside world
put an ACL on your DNS server to not let anyone outside your netblocks query
it... 2) resolv.conf on your fs box must point only at your name servers
that are SOA (either primary or secondary) for your private domain and much
not contain servers of your ISP that will cause lookup failures...
3) and once again you do not specify the IP address you are querying using
the enum command... You specify the number and the root (root being the
domain ie: mydomain.int)

Beyond that you need to check out other resources appropriate for setting up
enum records for your specific DNS implementation



From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 04 Dec 2008 16:12:36 +0700
To: "freeswitch-users at lists.freeswitch.org"
<freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch enum

Ken,

I have done what you suggested bellow, for security reason i can not paste
the query using the domain, just the localhost and i change the public  ip
to 1.2.3.4.
So i already setup a real name server with  real domain and it uses public
ip, configure the 7.6.5.4.3.2.1.e164.arpa as a domain with my name server as
SOA and NS record and yet the freeswitch CLI still give me no match when i
try to query it using my name server.

And yes, if i query 18005551212 e164.org it works like a charm ;-)

Any other suggestions?

Regards,

-Pieter-
I
Ken Rice wrote: 
>  
> Ok 1) Overriding the e164.arpa is probably not a good Idea... 2) the second
> param for the enum command is a domain not a Server IP address
> 
> Example: enum 1234567890 e164.org
> 
> You should probably set up your enum records on your own private domain or
> use a real domain that you own... If you have to use something like e164.int
> as the domain then tell your name server its the SOA for that domain and set
> all your records up there...
> 
> Then in FreeSwitch tell it the default domain is e164.int
> 
> Chances are if you do something like enum 18005551212 e164.org it will work
> correctly...
> 
> 
>   
>  
>>  
>> From: Pieter Eduard <pieter_eduard at biznetnetworks.com>
>> <mailto:pieter_eduard at biznetnetworks.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> Date: Thu, 04 Dec 2008 15:03:36 +0700
>> To: "freeswitch-users at lists.freeswitch.org"
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> <mailto:freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] freeswitch enum
>> 
>> Hi,
>> 
>> I'm trying to query freeswitch  to use my bind base enum server but i'm
>> having trouble to query the enum from Freeswitch CLI.
>> 
>> this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after
>> restart the dns, i have this :
>> 
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost
>> Using domain server:
>> Name: localhost
>> Address: 127.0.0.1#53
>> Aliases:
>> 
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 
>> and if use the ip, the result is the same :
>> [root at fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4
>> Using domain server:
>> Name: 1.2.3.4
>> Address: 1.2.3.4#53
>> Aliases:
>> 
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip"
>> "!^.*$!sip:1000 at test.com!" .
>> 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel"
>> "!^.*$!tel:\\12345678!" .
>> 
>> But if i try to query from the freeswitch CLI that's installed in the
>> same box, i get this :
>> 
>> freeswitch at fsbox> enum 12345678 localhost
>> API CALL [enum(12345678 localhost)] output:
>> No Match!
>> 
>> freeswitch at fsbox> enum 12345678 1.2.3.4
>> API CALL [enum(12345678 1.2.3.4)] output:
>> No Match!
>> 
>> even if i change the enum.conf.xml dns root with my enum ip and reload
>> the freeswitch i still get the same error.
>> 
>> Can anyone help me on this? i just want the fs to query on my local enum
>> or to query on different enum server.
>> 
>> Regards,
>> -Pieter-
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>>  
>>  
>> 
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> .
>> 
>>   
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

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From cstomi.levlist at gmail.com  Thu Dec  4 03:22:04 2008
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Thu, 04 Dec 2008 12:22:04 +0100
Subject: [Freeswitch-users] voicemail disk quota poll
Message-ID: <4937BD5C.7050002@gmail.com>

Hello,

I would like to use a disk quota in users' voicemail
(http://jira.freeswitch.org/browse/MODAPP-173)

We haven't decided yet what would be the better prompts to play to the 
caller when the mailbox is full.
Please advice some messages!

Thanks in advance,
Tamas





From faisalmaqsoodi at yahoo.com  Thu Dec  4 03:37:09 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Thu, 4 Dec 2008 03:37:09 -0800 (PST)
Subject: [Freeswitch-users] mod_spidermonkey()  Session is not active!
Message-ID: <966608.77509.qm@web30706.mail.mud.yahoo.com>


Hi
I m trying to use a Javascript application which uses? mod_pocketsphinx and mod_spidermonkey. Both r loaded but this error msg is displayed:
2008-12-04 16:31:09 [ERR] inline:1 mod_spidermonkey()? Session is not active!
How can i remove this error?
??????????????????????????????????? Faisal



      
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From saigop at gmail.com  Thu Dec  4 04:50:51 2008
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Thu, 4 Dec 2008 18:20:51 +0530
Subject: [Freeswitch-users] Predictive Dialing
Message-ID: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>

Hi,
   I would like to have predictive dialing. In asterisk we used manager api
and for outbound we use originate. The originate command will dial a number
where asterisk answer the call and then we predict the answering machine
with the silence file. Inspite of that human voice is detected and
transferred to agents.

   Like this how can we go ahead with Freeswitch? Any help would be
appreciated.

-- 
Thank you  with regards,
Gopal,
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From brian at freeswitch.org  Thu Dec  4 06:23:45 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2008 08:23:45 -0600
Subject: [Freeswitch-users] re gistration and calls through different
	user agent
In-Reply-To: <20829148.post@talk.nabble.com>
References: <20829148.post@talk.nabble.com>
Message-ID: <6643DA4A-019C-4446-8C5E-B19544187533@freeswitch.org>

Well this isn't how the default config is to be used.  The external  
profile is for talking to things outside your organization like  
gateways and service providers.  Phones shouldn't be registering to  
them unless you bond the internal and external profiles together.


In the internal profile you have:

     <!-- used to share presence info across sofia profiles -->
     <!-- Name of the db to use for this profile -->
     <!--<param name="dbname" value="share_presence"/>-->
     <!--<param name="presence-hosts" value="$${domain}"/>-->
     <!-- ************************************************* -->


Then on external you have these settings:

     <!-- Name of the db to use for this profile -->
     <!--<param name="dbname" value="share_presence"/>-->
     <!--<param name="presence-hosts" value="$${domain}"/>-->
     <!--<param name="force-register-domain" value="$${domain}"/>-->
     <!--all inbound reg will stored in the db using this domain -->
     <!--<param name="force-register-db-domain" value="$${domain}"/>-->
     <!-- ************************************************* -->


The settings will allow you to bind two profiles together and make  
them act as one.   That would be what you want I suspect.

/b

On Dec 4, 2008, at 2:28 AM, Carole O. wrote:

> Since both were able to register I guess that the directory
> /directory/default/ is used by both sofia agent else user1002 should  
> not be
> able to register, isn't? I am confused here because I would have  
> said that
> the directory /directory/default/ was just for the users that  
> register on
> internal...
>
> I had a second problem: if user1001 tries to call user1002 it did  
> not work
> (but the opposite is ok). Has this something to do with the dialplan?




From anthony.minessale at gmail.com  Thu Dec  4 07:34:46 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Dec 2008 09:34:46 -0600
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <1228365981.25709.60.camel@develop4>
References: <1228352588.25709.42.camel@develop4>
	<191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
	<1228365981.25709.60.camel@develop4>
Message-ID: <191c3a030812040734s4f514f42s9a30a48c93709fd5@mail.gmail.com>

most likely it's because during the time you are dong artificial ringback
the other side is not doing RTP right.

When the call is answered we flush the rtp buffer and your missing audio is
probably flushed with it.
so you can choose to have a 3 second delay or erase the 3 seconds as it does
now.

This is a known problem with sonus which has been proven to build up an
audio delay during the time
you are waiting for the call to answer.  I'm sure you prefer the way it is
to a large audio delay.



On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <ack at telefonica.net>wrote:

> No TDM , all is SIP :
>
>
> PSTN ---> Sip Proxy_A --> FS ( brigde ) ringback/transfer_ringback
> -> Sip Proxy_B --> PSTN
>
>
> In logfile i think you can get some details about Media Gateways
> ( Sonus ) PSTN inbound / outbound is provided by Level3.
>
> I can get a capture of a call if you want, in capture the audio is not
> missing, issue with :
>
> - rtp buffer ?
> - Sonus ?
>
> Let me know anything you need so i can provide a log or create a new
> scenario.
>
>
> Thanks,
>
> El mi?, 03-12-2008 a las 22:12 -0600, Anthony Minessale escribi?:
> > what does PSTN represent?
> >
> > I know what the PSTN is but how are you reaching it?
> > is it TDM, SIP etc... what gateway type other details.
> >
> >
> > On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <ack at telefonica.net>
> > wrote:
> >         Hi guys,
> >
> >          I've a strange issue with FS , version svn -r10584 ,
> >         when FS bridges a call first 3 seconds of audio are missing ,
> >         looks that
> >         only happens on PSTN calls and using ringback or
> >         transfer_ringback. This
> >         only happens in calls from PSTN , not from VOIP. Some
> >         scenarios i tried
> >         to isolate this issue :
> >
> >
> >         - Issue
> >
> >         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
> >
> >         - Good setting bypass_media before run bridge but i need rtp
> >         in FS path
> >
> >         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
> >
> >         - Good
> >
> >         PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback ->
> >         PSTN
> >
> >         - Always good
> >
> >         VOIP --> FS ( brigde ) -> PSTN
> >
> >
> >         Dialplan has nothing wrong ( i guess ):
> >
> >         <extension name="Transfers">
> >            <condition field="destination_number"
> >         expression="^1??XXXXXXXXXX$">
> >              <action application="answer"/>
> >              <action application="speak" data="cepstral|Allison-8kHz|
> >         blah"/>
> >              <action application="set"
> >         data="hangup_after_bridge=false"/>
> >              <action application="set" data="playback_terminators=#"/>
> >              <action application="set" data="ringback=$${us-ring}"/>
> >              <action application="set" data="transfer_ringback=
> >         $${hold_music}"/>
> >              <action application="set" data="effective_caller_id_name=
> >         ${caller_id_name}"/>
> >              <action application="set"
> >         data="effective_caller_id_number=
> >         ${caller_id_number}"/>
> >              <action application="set" data="originate_timeout=30"/>
> >              <action application="set" data="call_timeout=30"/>
> >              <action application="bridge"
> >         data="sofia/default/18008226235 at PSTN_GW"/>
> >              <action application="speak" data="cepstral|Allison-8kHz|
> >         Transfer
> >         finished"/>
> >              <action application="hangup"/>
> >            </condition>
> >          </extension>
> >
> >
> >
> >         Any ideas ?
> >
> >         Attached log of FS ( F8 from console ).
> >
> >
> >         Thanks in advance !
> >
> >         --
> >         Angel Carpintero
> >         ack ( at ) telefonica ( dot ) net
> >
> >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
> >         6EF1 B90D
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
>
> --
> Angel Carpintero
> ack ( at ) telefonica ( dot ) net
>
> Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Dec  4 08:07:36 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 4 Dec 2008 08:07:36 -0800
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
Message-ID: <F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>

Gopal,

FreeSWITCH does not have free amd but you can buy a license. Please  
send a request to consulting at freeswitch.org. I have some experience  
with running amd so I can assist with setup questions.

-Michael

Sent from my iPhone

On Dec 4, 2008, at 4:50 AM, "Gopalakrishnan A.N" <saigop at gmail.com>  
wrote:

> Hi,
>
>    I would like to have predictive dialing. In asterisk we used  
> manager api and for outbound we use originate. The originate command  
> will dial a number where asterisk answer the call and then we  
> predict the answering machine with the silence file. Inspite of that  
> human voice is detected and transferred to agents.
>
>    Like this how can we go ahead with Freeswitch? Any help would be  
> appreciated.
>
> -- 
> Thank you  with regards,
> Gopal,
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gkuri at ieee.org  Thu Dec  4 09:25:56 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 04 Dec 2008 09:25:56 -0800
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <4937BD5C.7050002@gmail.com>
References: <4937BD5C.7050002@gmail.com>
Message-ID: <493812A4.4080404@ieee.org>

How about:

"The mailbox of the person you are trying to reach is full and can not
accept new messages at this time. Please try your call again later. Goodbye"

~Gabe

Tamas Cseke wrote:
> Hello,
> 
> I would like to use a disk quota in users' voicemail
> (http://jira.freeswitch.org/browse/MODAPP-173)
> 
> We haven't decided yet what would be the better prompts to play to the
> caller when the mailbox is full.
> Please advice some messages!
> 
> Thanks in advance,
> Tamas
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From msc at freeswitch.org  Thu Dec  4 09:43:26 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Dec 2008 09:43:26 -0800
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <493812A4.4080404@ieee.org>
References: <4937BD5C.7050002@gmail.com> <493812A4.4080404@ieee.org>
Message-ID: <87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>

Nice! I'll add that to my list of new prompts to be recorded. FYI, if
you have any other suggestions please email this list or post a
comment here:

http://jira.freeswitch.org/browse/FSSCRIPTS-9
-MC

On Thu, Dec 4, 2008 at 9:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
>
> How about:
>
> "The mailbox of the person you are trying to reach is full and can not
> accept new messages at this time. Please try your call again later. Goodbye"
>
> ~Gabe
>
> Tamas Cseke wrote:
> > Hello,
> >
> > I would like to use a disk quota in users' voicemail
> > (http://jira.freeswitch.org/browse/MODAPP-173)
> >
> > We haven't decided yet what would be the better prompts to play to the
> > caller when the mailbox is full.
> > Please advice some messages!
> >
> > Thanks in advance,
> > Tamas
> >
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Thu Dec  4 09:50:36 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Dec 2008 11:50:36 -0600
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>
References: <4937BD5C.7050002@gmail.com> <493812A4.4080404@ieee.org>
	<87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>
Message-ID: <191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>

maybe
http://www.sofaswitch.org/eg/sounds/fucked.wav



On Thu, Dec 4, 2008 at 11:43 AM, Michael Collins <msc at freeswitch.org> wrote:

> Nice! I'll add that to my list of new prompts to be recorded. FYI, if
> you have any other suggestions please email this list or post a
> comment here:
>
> http://jira.freeswitch.org/browse/FSSCRIPTS-9
> -MC
>
> On Thu, Dec 4, 2008 at 9:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
> >
> > How about:
> >
> > "The mailbox of the person you are trying to reach is full and can not
> > accept new messages at this time. Please try your call again later.
> Goodbye"
> >
> > ~Gabe
> >
> > Tamas Cseke wrote:
> > > Hello,
> > >
> > > I would like to use a disk quota in users' voicemail
> > > (http://jira.freeswitch.org/browse/MODAPP-173)
> > >
> > > We haven't decided yet what would be the better prompts to play to the
> > > caller when the mailbox is full.
> > > Please advice some messages!
> > >
> > > Thanks in advance,
> > > Tamas
> > >
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Dec  4 09:55:43 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Dec 2008 09:55:43 -0800
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>
References: <4937BD5C.7050002@gmail.com> <493812A4.4080404@ieee.org>
	<87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>
	<191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>
Message-ID: <87f2f3b90812040955h243bd649gdbca184c35815b96@mail.gmail.com>

I think GM Voices levies a "naughtiness surcharge" but I'll see what I
can find out. :)
-MC

On Thu, Dec 4, 2008 at 9:50 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> maybe
> http://www.sofaswitch.org/eg/sounds/fucked.wav
>
>
>
> On Thu, Dec 4, 2008 at 11:43 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> Nice! I'll add that to my list of new prompts to be recorded. FYI, if
>> you have any other suggestions please email this list or post a
>> comment here:
>>
>> http://jira.freeswitch.org/browse/FSSCRIPTS-9
>> -MC
>>
>> On Thu, Dec 4, 2008 at 9:25 AM, Gabriel Kuri <gkuri at ieee.org> wrote:
>> >
>> > How about:
>> >
>> > "The mailbox of the person you are trying to reach is full and can not
>> > accept new messages at this time. Please try your call again later.
>> > Goodbye"
>> >
>> > ~Gabe
>> >
>> > Tamas Cseke wrote:
>> > > Hello,
>> > >
>> > > I would like to use a disk quota in users' voicemail
>> > > (http://jira.freeswitch.org/browse/MODAPP-173)
>> > >
>> > > We haven't decided yet what would be the better prompts to play to the
>> > > caller when the mailbox is full.
>> > > Please advice some messages!
>> > >
>> > > Thanks in advance,
>> > > Tamas
>> > >
>> > >
>> > >
>> > > _______________________________________________
>> > > Freeswitch-users mailing list
>> > > Freeswitch-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Thu Dec  4 10:03:20 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2008 12:03:20 -0600
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <87f2f3b90812040955h243bd649gdbca184c35815b96@mail.gmail.com>
References: <4937BD5C.7050002@gmail.com> <493812A4.4080404@ieee.org>
	<87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>
	<191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>
	<87f2f3b90812040955h243bd649gdbca184c35815b96@mail.gmail.com>
Message-ID: <076D0E41-DE7A-494C-8096-352BBE104246@freeswitch.org>

Actually GM Voices won't do anything with profanity in it.

/b

On Dec 4, 2008, at 11:55 AM, Michael Collins wrote:

> I think GM Voices levies a "naughtiness surcharge" but I'll see what I
> can find out. :)
> -MC




From odermann at googlemail.com  Thu Dec  4 11:45:36 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 4 Dec 2008 20:45:36 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
Message-ID: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>

hi,

after we managed to setup fs with mod_fax and our socket outbound
script, we have some questioons about an error and problems, when
sending a fax:

1.) there is one error, we get always - no matter, if the fax was sent
successfully or not.
in the pastebin under http://pastebin.freeswitch.org/6338 you can see
the error in the last line.
this is the full log of a fax in fs console loglevel debug.


2.) fax works quite good. we couls send long faxes over a normal fax
machine without any problem.
but for fast testing we are using a softfax (fritz fax). here we have
some more problems.
mostly a fax with one page will pass through, but more pages will mostly fail.
because we are new to the fax thing, we do not really know, what the
messages tell us about failed faxes.
here are the top 3 messages (unordered and always one at a time -
nerver at once):

variable_fax_result_text => Received a DCN while waiting for a DIS

fax_result_text => The HDLC carrier did not stop in a timely manner

fax_result_text => Unexpected message received


could someone please tell us, where the problem might be?


thanks
dennis



From msc at freeswitch.org  Thu Dec  4 12:09:21 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Dec 2008 12:09:21 -0800
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
Message-ID: <87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>

Dennis,

Thanks for your input on the fax stuff! We will check this out and report back.

Question: if one of the devs would like to SSH into your system to do
further testing, is that okay?

Thanks,
MC

On Thu, Dec 4, 2008 at 11:45 AM, Dennis <odermann at googlemail.com> wrote:
> hi,
>
> after we managed to setup fs with mod_fax and our socket outbound
> script, we have some questioons about an error and problems, when
> sending a fax:
>
> 1.) there is one error, we get always - no matter, if the fax was sent
> successfully or not.
> in the pastebin under http://pastebin.freeswitch.org/6338 you can see
> the error in the last line.
> this is the full log of a fax in fs console loglevel debug.
>
>
> 2.) fax works quite good. we couls send long faxes over a normal fax
> machine without any problem.
> but for fast testing we are using a softfax (fritz fax). here we have
> some more problems.
> mostly a fax with one page will pass through, but more pages will mostly fail.
> because we are new to the fax thing, we do not really know, what the
> messages tell us about failed faxes.
> here are the top 3 messages (unordered and always one at a time -
> nerver at once):
>
> variable_fax_result_text => Received a DCN while waiting for a DIS
>
> fax_result_text => The HDLC carrier did not stop in a timely manner
>
> fax_result_text => Unexpected message received
>
>
> could someone please tell us, where the problem might be?
>
>
> thanks
> dennis
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Thu Dec  4 12:17:40 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2008 14:17:40 -0600
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
Message-ID: <195B4EA0-8548-4FA7-8B72-DE2315D69F7B@freeswitch.org>

Also need to know is this via SIP or TDM?

/b

On Dec 4, 2008, at 2:09 PM, Michael Collins wrote:

> Dennis,
>
> Thanks for your input on the fax stuff! We will check this out and  
> report back.
>
> Question: if one of the devs would like to SSH into your system to do
> further testing, is that okay?
>
> Thanks,
> MC




From brian at freeswitch.org  Thu Dec  4 12:17:40 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2008 14:17:40 -0600
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
Message-ID: <195B4EA0-8548-4FA7-8B72-DE2315D69F7B@freeswitch.org>

Also need to know is this via SIP or TDM?

/b

On Dec 4, 2008, at 2:09 PM, Michael Collins wrote:

> Dennis,
>
> Thanks for your input on the fax stuff! We will check this out and  
> report back.
>
> Question: if one of the devs would like to SSH into your system to do
> further testing, is that okay?
>
> Thanks,
> MC




From odermann at googlemail.com  Thu Dec  4 12:25:14 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 4 Dec 2008 21:25:14 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<87f2f3b90812041209s79a753a2m9fe8f703b3651ba9@mail.gmail.com>
Message-ID: <5e414ed0812041225x4efa682ev17b1a366ecfd5654@mail.gmail.com>

my server (including me) is your slave. anthony already feels home on
my server, so you are greatly invited ;-)

it is quite late in germany, so i feel, that we should meet in irc
tomorrow, if this is ok for you.


@brian
we get everthing over sip. so we receive the faxes over sip. the
faxes, which we send for testing (not over fs or the same machine) are
sent over isdn.


2008/12/4 Michael Collins <msc at freeswitch.org>:
> Dennis,
>
> Thanks for your input on the fax stuff! We will check this out and report back.
>
> Question: if one of the devs would like to SSH into your system to do
> further testing, is that okay?



From steveu at coppice.org  Thu Dec  4 16:23:33 2008
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 05 Dec 2008 08:23:33 +0800
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
Message-ID: <49387485.9000303@coppice.org>

Dennis wrote:
> hi,
>
> after we managed to setup fs with mod_fax and our socket outbound
> script, we have some questioons about an error and problems, when
> sending a fax:
>
> 1.) there is one error, we get always - no matter, if the fax was sent
> successfully or not.
> in the pastebin under http://pastebin.freeswitch.org/6338 you can see
> the error in the last line.
> this is the full log of a fax in fs console loglevel debug.
>   
That looks like something annoying but harmless. It looks like the comms 
path is disabled slightly before the flow of packets is turned off. 
That's probably just a silly slip in the code.
>
> 2.) fax works quite good. we couls send long faxes over a normal fax
> machine without any problem.
> but for fast testing we are using a softfax (fritz fax). here we have
> some more problems.
> mostly a fax with one page will pass through, but more pages will mostly fail.
> because we are new to the fax thing, we do not really know, what the
> messages tell us about failed faxes.
> here are the top 3 messages (unordered and always one at a time -
> nerver at once):
>
> variable_fax_result_text => Received a DCN while waiting for a DIS
>
> fax_result_text => The HDLC carrier did not stop in a timely manner
>
> fax_result_text => Unexpected message received
>
>
> could someone please tell us, where the problem might be?
>   
Does Fritz FAX means the ISDN card stuff? If so, that should be 
something well proven. However, the errors you are getting sound like 
the FAX at the far end is buggy. I think a log of the audio from one or 
two of these calls is needed for analysis.

Regards,
Steve




From dave at 3c.co.uk  Fri Dec  5 00:15:37 2008
From: dave at 3c.co.uk (David Knell)
Date: Fri, 05 Dec 2008 08:15:37 +0000
Subject: [Freeswitch-users] voicemail disk quota poll
In-Reply-To: <076D0E41-DE7A-494C-8096-352BBE104246@freeswitch.org>
References: <4937BD5C.7050002@gmail.com>
	<493812A4.4080404@ieee.org>	<87f2f3b90812040943m2bb461afh7de256a49db710aa@mail.gmail.com>	<191c3a030812040950r57139e11k29186aa7dc1aee5f@mail.gmail.com>	<87f2f3b90812040955h243bd649gdbca184c35815b96@mail.gmail.com>
	<076D0E41-DE7A-494C-8096-352BBE104246@freeswitch.org>
Message-ID: <4938E329.9030908@3c.co.uk>

I still know some folk in the 900-number business.  They won't do 
anything without
profanity in it ;-)

--Dave
> Actually GM Voices won't do anything with profanity in it.
>
> /b
>
> On Dec 4, 2008, at 11:55 AM, Michael Collins wrote:
>
>   
>> I think GM Voices levies a "naughtiness surcharge" but I'll see what I
>> can find out. :)
>> -MC
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From faisalmaqsoodi at yahoo.com  Fri Dec  5 00:37:24 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Fri, 5 Dec 2008 00:37:24 -0800 (PST)
Subject: [Freeswitch-users] Handling directory of sound files
Message-ID: <228017.80650.qm@web30704.mail.mud.yahoo.com>

Hi,
?Can i accomplish folder tasks with freeswitch? For instance, i need to play all sound files contained in a directory sequentially or randomly. Plz help me doing this.
????????????????????????????????????????????????????????????????????????????????????????????????? Faisal



      
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From msc at freeswitch.org  Fri Dec  5 00:36:09 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 00:36:09 -0800
Subject: [Freeswitch-users] Nice FS article
Message-ID: <87f2f3b90812050036n72d85fbcqf24a2d1f0c878a8d@mail.gmail.com>

Check it out:
http://digg.com/software/FreeSWITCH_knocks_Asterisk_s_block_off

Please diggit left and right!!

-MC



From msc at freeswitch.org  Fri Dec  5 00:47:04 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 00:47:04 -0800
Subject: [Freeswitch-users] Handling directory of sound files
In-Reply-To: <228017.80650.qm@web30704.mail.mud.yahoo.com>
References: <228017.80650.qm@web30704.mail.mud.yahoo.com>
Message-ID: <87f2f3b90812050047i73e34e3u10f9c1b4b1fb4704@mail.gmail.com>

Is this for Music on Hold? Or is it for a different application altogether?
Thanks,
MC

On Fri, Dec 5, 2008 at 12:37 AM, Faisal Maqsoodi
<faisalmaqsoodi at yahoo.com> wrote:
> Hi,
>  Can i accomplish folder tasks with freeswitch? For instance, i need to play
> all sound files contained in a directory sequentially or randomly. Plz help
> me doing this.
>
> Faisal
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From faisalmaqsoodi at yahoo.com  Fri Dec  5 00:59:48 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Fri, 5 Dec 2008 00:59:48 -0800 (PST)
Subject: [Freeswitch-users] Handling directory of sound files
In-Reply-To: <87f2f3b90812050047i73e34e3u10f9c1b4b1fb4704@mail.gmail.com>
Message-ID: <692105.73475.qm@web30706.mail.mud.yahoo.com>

Its not without music on hold completely. Say, e.g, moh is being played but when i press 1 it should start playing files contained in a specific directory sequentially or randomly. Hope i m able to explain.
? ? ? ? ? ? ? ? ? ? ? ?? ???????????????????????? Faisal

--- On Fri, 12/5/08, Michael Collins <msc at freeswitch.org> wrote:
From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Handling directory of sound files
To: freeswitch-users at lists.freeswitch.org
Date: Friday, December 5, 2008, 12:47 AM

Is this for Music on Hold? Or is it for a different application altogether?
Thanks,
MC

On Fri, Dec 5, 2008 at 12:37 AM, Faisal Maqsoodi
<faisalmaqsoodi at yahoo.com> wrote:
> Hi,
>  Can i accomplish folder tasks with freeswitch? For instance, i need to
play
> all sound files contained in a directory sequentially or randomly. Plz
help
> me doing this.
>
> Faisal
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From mrjoebain at gmail.com  Fri Dec  5 01:10:23 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Fri, 5 Dec 2008 09:10:23 +0000
Subject: [Freeswitch-users] Javascript ODBC on Windows
In-Reply-To: <E05E9383-0BF2-4B8B-93A5-09FB99817F19@jerris.com>
References: <748d53500812020629p6a0d178dh672cec871c018254@mail.gmail.com>
	<E05E9383-0BF2-4B8B-93A5-09FB99817F19@jerris.com>
Message-ID: <748d53500812050110u6fffe498tb3fe6ff5b64600af@mail.gmail.com>

Thanks, you're right it seems to be an odbc problem, 64bit / 32bit clash I
think.

Joe

2008/12/2 Michael Jerris <mike at jerris.com>

> Yes, it should work fine.  As the error message says it didn't find
> the data source name you specified.  You need to setup your odbc data
> source on the system
>
> Mike
>
> On Dec 2, 2008, at 9:29 AM, Joe Bain wrote:
>
> > Hi all,
> >
> > Is it possible to use mod_spidermonkey_odbc with a Windows
> > installation of FreeSWITCH at the moment? If so does anyone have any
> > pointers? I get:
> >
> > 2008-12-02 14:23:57 [DEBUG] switch_odbc.c:145
> > switch_odbc_handle_connect() Connecting ivr_test
> > 2008-12-02 14:23:57 [ERR] switch_odbc.c:160
> > switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [Microsoft]
> > [ODBC Driver Manager] Data source name not found and no default
> > driver specified
> >
> > when I try.
> >
> > Thanks in advance,
> >
> > Joe Bain
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mrjoebain at gmail.com  Fri Dec  5 02:33:42 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Fri, 5 Dec 2008 10:33:42 +0000
Subject: [Freeswitch-users] Problem reloading xml
Message-ID: <748d53500812050233y3b965e13ofcb566a1e83bbb14@mail.gmail.com>

Hi all,

I have come across a strange problem when using the phrases in conf/lang/en.
Initially I had a problem where FreeSwitch wouldn't load new subdirectories,
even when I included their paths in the en.xml file. I went ahead writing
all the phrases (I only have 3 so far) in en.xml but now it doesn't
recognise even these properly. I thought there might be some conflicts with
the naming of phrases so I deleted all the included demo phrases and
commented their lines out of the en.xml file but when I do 'reloadxml' I get
the error:

Error including C:\Program Files (x86)\FreeSWITCH\conf\lang\en\demo/*.xml
(No su
ch file or directory)
Error including C:\Program Files (x86)\FreeSWITCH\conf\lang\en\test/*.xml
(No su
ch file or directory)
Error including C:\Program Files (x86)\FreeSWITCH\conf\lang\en\vm/sounds.xml
(No
 such file or directory)

This persists after restarting FS too. Very confusing, any ideas? I am
running FS on Vista 64 bit, but with the 32 bit version. The version of FS
is "FreeSWITCH Version 1.0.trunk (10175M)".

Thanks

Joe
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From odermann at googlemail.com  Fri Dec  5 02:54:02 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 5 Dec 2008 11:54:02 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <49387485.9000303@coppice.org>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
Message-ID: <5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>

2008/12/5 Steve Underwood <steveu at coppice.org>:
>> 1.) there is one error, we get always - no matter, if the fax was sent
>> successfully or not.
>> in the pastebin under http://pastebin.freeswitch.org/6338 you can see
>> the error in the last line.
>> this is the full log of a fax in fs console loglevel debug.
>>
> That looks like something annoying but harmless. It looks like the comms
> path is disabled slightly before the flow of packets is turned off.
> That's probably just a silly slip in the code.

yup, because the error always appear, i successful or not, this error
can't be a big problem. i just do not like red lines in the log ;-)


> Does Fritz FAX means the ISDN card stuff? If so, that should be
> something well proven. However, the errors you are getting sound like
> the FAX at the far end is buggy. I think a log of the audio from one or
> two of these calls is needed for analysis.

yes, fritz fax is the isdn stuff. normally it works very well.
how can i get a log of the audio? when a fax is coming in, there
happens quite little in the console at loglevel debug. i pasted all
into the pastebin.
if there are more possibilities to get mor information, please let me know.



From jan.kubr at gmail.com  Fri Dec  5 03:08:24 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Fri, 5 Dec 2008 12:08:24 +0100
Subject: [Freeswitch-users] DTMF from cell phones
Message-ID: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>

Hi,
recently someone was mentioning an issue with DTMF here, but there was
no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
(read app doesn't terminate). I could not find any regularity in this,
sometimes it is recognized just fine, sometimes I had to wait for the
file to be played etc. The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is
recognized perfectly. So it's probably related to GSM or something.

I was wondering whether anyone experienced the same and whether there
is something I can do about it. There are a few DTMF-related variables
in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
them a bit, but I don't really know what I'm doing.. Couldn't find any
docs, either.
Any ideas would be appreciated.

Jan Kubr



From saigop at gmail.com  Fri Dec  5 03:23:36 2008
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Fri, 5 Dec 2008 16:53:36 +0530
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
	<F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
Message-ID: <2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>

Hi Micheal,
  Thanks for the reply! cant I try with tone detect?

  Like dial a number in session and try to detect with tone detect and then
bridge the call with some extension.

-- 
Thank you  with regards,
Gopal,
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From faisalmaqsoodi at yahoo.com  Fri Dec  5 03:48:13 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Fri, 5 Dec 2008 03:48:13 -0800 (PST)
Subject: [Freeswitch-users] Handling directory of sound files
Message-ID: <132465.82190.qm@web30706.mail.mud.yahoo.com>

Its not without music on hold completely. Say, e.g, moh is being played
but when i press 1 it should start playing files contained in a
specific directory sequentially or randomly. I havent got any solution to this problem yet. Can anyone plz guide me to some documentation or anything else regarding this matter.
? ? ? ? ? ? ? ? ? ? ? ?? ???????????????????? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? Faisal


      
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From Prometheus001 at gmx.net  Fri Dec  5 03:54:54 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Dec 2008 12:54:54 +0100
Subject: [Freeswitch-users] Event_Socket: How to determine the ogininater
 uuid before an outgoing call is set up?
Message-ID: <4939168E.9020400@gmx.net>

I am building an IVR application where an incoming call is initiating an
outgoing call. When I pass a "variable_other_uuid" (the uuid of the
incoming channel) at originate time, I am able to reference to the
incomig call, once the outgoing call is set up. So the outgoing call can
see the uuid of the incoming call who has originated the outgoing call.
This is needed for bridging the 2 calls together.

However I want to control also the call setup process (see, if the
outgoing call is ringing etc.). At call setup time, when I parse the
channel_originate ,channel_outgoing and channel_progress events, I
cannot see any reference to the incoming call (variable_other_uuid is
not set). I suspect that variables are only passed once the outgoing
channel is set up.

Has anybody an idea, how I may get the uuid of the originating uuid in
the outgoing call at call setup?

Best regards
Peter



From regs at kinetix.gr  Fri Dec  5 04:23:28 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 14:23:28 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
Message-ID: <49391D40.6050103@kinetix.gr>

The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this nornal?

And another question : what piece of info could inform me about who 
hanged up?


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 




From ack at telefonica.net  Fri Dec  5 04:30:25 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Fri, 05 Dec 2008 13:30:25 +0100
Subject: [Freeswitch-users] DTMF from cell phones
In-Reply-To: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>
References: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>
Message-ID: <1228480225.25709.100.camel@develop4>

I had some issues with some previous versions of FS , in trunk looks
that is fixed. ( Notice current svn revision is 10609 )

in sip profiles i have :

...
<param name="rfc2833-pt" value="101"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${global_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="none"/>
<param name="inbound-codec-negotiation" value="greedy"/>
...

As codecs g711 ULAW (PCMU):

in vars.xml.conf :

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM"/>

So i guess that using latest version with a few changes in your config 
should work unless there's any other issue related to your sip provider
( PSTN / Media Gateway ), on this case you can get some captures of
sip/rtp traffic to check SDP and rtp Marks.

El vie, 05-12-2008 a las 12:08 +0100, Jan Kubr escribi?:
> Hi,
> recently someone was mentioning an issue with DTMF here, but there was
> no solution. I have a similar problem, when calling Freeswitch from my
> cell phone (via a SIP provider), sometimes DTMF is not recognized
> (read app doesn't terminate). I could not find any regularity in this,
> sometimes it is recognized just fine, sometimes I had to wait for the
> file to be played etc. The important thing to note is that when using
> a SIP softphone (X-Lite) I have never had this problem, DTMF is
> recognized perfectly. So it's probably related to GSM or something.
> 
> I was wondering whether anyone experienced the same and whether there
> is something I can do about it. There are a few DTMF-related variables
> in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
> dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
> them a bit, but I don't really know what I'm doing.. Couldn't find any
> docs, either.
> Any ideas would be appreciated.
> 
> Jan Kubr
> 

Cheers,
-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

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From mike at jerris.com  Fri Dec  5 06:39:14 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 09:39:14 -0500
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
Message-ID: <3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>


On Dec 5, 2008, at 5:54 AM, Dennis wrote:

> 2008/12/5 Steve Underwood <steveu at coppice.org>:
>>> 1.) there is one error, we get always - no matter, if the fax was  
>>> sent
>>> successfully or not.
>>> in the pastebin under http://pastebin.freeswitch.org/6338 you can  
>>> see
>>> the error in the last line.
>>> this is the full log of a fax in fs console loglevel debug.
>>>
>> That looks like something annoying but harmless. It looks like the  
>> comms
>> path is disabled slightly before the flow of packets is turned off.
>> That's probably just a silly slip in the code.
>
> yup, because the error always appear, i successful or not, this error
> can't be a big problem. i just do not like red lines in the log ;-)
>
>
>> Does Fritz FAX means the ISDN card stuff? If so, that should be
>> something well proven. However, the errors you are getting sound like
>> the FAX at the far end is buggy. I think a log of the audio from  
>> one or
>> two of these calls is needed for analysis.
>
> yes, fritz fax is the isdn stuff. normally it works very well.
> how can i get a log of the audio? when a fax is coming in, there
> happens quite little in the console at loglevel debug. i pasted all
> into the pastebin.
> if there are more possibilities to get mor information, please let  
> me know.

You should be able to record session with stereo to get the audio  
here, the other easy way would be to use pcapsipdump tool (look for it  
on the wiki) to get a pcap of the whole call.

Mike




From mike at jerris.com  Fri Dec  5 06:40:55 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 09:40:55 -0500
Subject: [Freeswitch-users] DTMF from cell phones
In-Reply-To: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>
References: <698401620812050308j61dda70bq5669daa5b9282aa5@mail.gmail.com>
Message-ID: <58C18F16-E8A3-4963-A624-32BFE13D2C26@jerris.com>


On Dec 5, 2008, at 6:08 AM, Jan Kubr wrote:

> Hi,
> recently someone was mentioning an issue with DTMF here, but there was
> no solution. I have a similar problem, when calling Freeswitch from my
> cell phone (via a SIP provider), sometimes DTMF is not recognized
> (read app doesn't terminate). I could not find any regularity in this,
> sometimes it is recognized just fine, sometimes I had to wait for the
> file to be played etc. The important thing to note is that when using
> a SIP softphone (X-Lite) I have never had this problem, DTMF is
> recognized perfectly. So it's probably related to GSM or something.
>
> I was wondering whether anyone experienced the same and whether there
> is something I can do about it. There are a few DTMF-related variables
> in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt,
> dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with
> them a bit, but I don't really know what I'm doing.. Couldn't find any
> docs, either.
> Any ideas would be appreciated.

If it is coming from the sip provider as rfc 2833 dtmf, they are  
probably doing inband detection and failing at it.  If you look at an  
rtp dump you can confirm this.  If this is the case, there is nothing  
you can do on the FreeSWITCH side and the provider will have to fix it.

Mike



From mike at jerris.com  Fri Dec  5 06:41:29 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 09:41:29 -0500
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
	<F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
	<2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>
Message-ID: <DB682CA7-D304-4D78-ABA7-4BD91FF6A8AA@jerris.com>


On Dec 5, 2008, at 6:23 AM, Gopalakrishnan A.N wrote:

> Hi Micheal,
>
>   Thanks for the reply! cant I try with tone detect?
>
>   Like dial a number in session and try to detect with tone detect  
> and then bridge the call with some extension.

If you know the exact frequency of the tone you can, but I suspect you  
do not.

Mike




From mike at jerris.com  Fri Dec  5 06:42:38 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 09:42:38 -0500
Subject: [Freeswitch-users] Event_Socket: How to determine the
	ogininater uuid before an outgoing call is set up?
In-Reply-To: <4939168E.9020400@gmx.net>
References: <4939168E.9020400@gmx.net>
Message-ID: <41CC850A-05A4-4618-B413-A655A7CCEA38@jerris.com>


On Dec 5, 2008, at 6:54 AM, Peter P GMX wrote:

> I am building an IVR application where an incoming call is  
> initiating an
> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
> incoming channel) at originate time, I am able to reference to the
> incomig call, once the outgoing call is set up. So the outgoing call  
> can
> see the uuid of the incoming call who has originated the outgoing  
> call.
> This is needed for bridging the 2 calls together.
>
> However I want to control also the call setup process (see, if the
> outgoing call is ringing etc.). At call setup time, when I parse the
> channel_originate ,channel_outgoing and channel_progress events, I
> cannot see any reference to the incoming call (variable_other_uuid is
> not set). I suspect that variables are only passed once the outgoing
> channel is set up.

Control in what way?

>
>
> Has anybody an idea, how I may get the uuid of the originating uuid in
> the outgoing call at call setup?




From zolotov at altron.ua  Fri Dec  5 06:53:05 2008
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Fri, 05 Dec 2008 16:53:05 +0200
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
Message-ID: <1228488785.5481.4.camel@opos20.altron.lan>

  Greetings!  

Question about possibility of the use FreeSWITCH for work with T1/E1
streams under Sun Solaris 10 a bit clears up (Solaris 11 is in condition
of alpha-version and not suitable for the industrial use). But answers
carry more negative sense.

Start of T1/E1 under Sun Solaris has 2 stages: a) start of wanpipe's
interface; b) make FreeSWITCH for Sun Solaris.


a) For Linux Sangoma recommends
(http://wiki.sangoma.com/wanpipe-freeswitch-install) installation of new
interface wanpipe-3.3.14.tgz, which has beta-status and placed at
ftp://ftp.sangoma.com/linux/current_wanpipe.

This installation and tests were successful.

In this release (2007 y.) by Sangoma was added TDM API - native
interface for FreeSwitch (YATE) ? which is absent in all previous
releases of WANPIPE. In the Linux release 3.3.14 this variant was named
as ?TDM API?, in older releases - ?TDM Voice?.

In this case FreeSWITCH works very good with E1/T1 cards
A-101/102/104/108 without installation in zaptel system, and spans in
configuration files of FreeSWITCH declared as [wanpipe#].

But for Sun Solaris we found only
drverftp://ftp.sangoma.com/Solaris/Beta/SVwanpipe-i386-5.10.pkgbeta,
which dated 2007:

> NAME=sangoma.com Wanpipe Driver

> VERSION=1.1.0,REV=2007-07-16,

Packet Svwanpipe-i386-5.10.pkg uses Svzaptel-i386-5.10.pkg, which wasn't
developed by Sangoma, but by little free
communityhttp://www.solarisvoip.com/, source codes of this packet is
herehttps://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk/

The analysis of source codes shows that this project develops very
slowly (there are no updates for about year), has very limited
functionality and supports ( unlike original zaptel ) very limited list
of cards ( only one ;) - Digium Wildcard TE110P T1/PRI).

Svwanpipe-i386-5.10.pkg supports only 64-bit Sun Solaris (on CD, which
we get with Sangoma's cards, presents 32-bit driver and PDF document
about installation under Sun Solaris ? but it dated 2001 ? 2002 yy).

At first we have checked up installation with TDM Voice + zaptel (like
for Asterisk) under Linux. We configured PRI spans as [zt] ... - such
installation works good and we could do calls :


> originate openzap/1/A/20000 &sleep(3)


      * evidently that call retranslates from span 1 to span 2,
        connected with cross-cable, and goes to extension 20000.
        


b) About making FreeSwitch:

Under Sun Solaris 10 with GCC makes FreeSWITCH core and most modules,
except openzapand some others, because Sun Solaris 10 has GCC 3.4,
but ./configure for openzap requires compatibility with ANSII 99.

We have checked 3 different methods of makinf FreeSWITCH :


1. set up GCC 4.0.2 from CSW-repositaries (and all reguired for GCC
*.pkg);

2. set up SunStudio 12 and do make cc/CC; 

3.cross-compile FreeSwitch for Sun Solaris under Linux with CC-options.


In testing we used assembly SunStudio as 32-bit application (64-bit
comes to the end with mistakes of assembly of some libraries, it is
possible to correct for it easily, but we did not begin to specify it).


For testing cards A-104 we have repeated the same installation and
configuring as under Linux (wanpipe TDM Voice + zaptel) on 4 different
64-bit servers under Sun Solaris 10:


      * on 3 servers (2 of them manufactured by SUN) executed
        successfully : # wanrouter start
        


4 wanrouter# interfaces was created; leds on spans, connected with
cross-cable, becomes GREEN, i.e. synchronization T1/E1 presents (no
alarms).

On these servers FreeSWITCH correctly makes:


> load mod_openzap


      * but when we make :
        


> originate openzap/1/A/20000 &sleep(3)


- for 2 connected with crosss-cable spans (1 & 2) FreeSWITCH transmits
PRI message (chan 1/31), but chan 2/31 doesn't receive this message
(unlike under Linux) and call breaks after timeout.

On 4-th server wanpipe doesn't even starts by mistake some IOCTL (i.e.
at a command for device). Three ?working? servers is on AMD Opteron Dual
Core 2214(F), fourth is on ntel Xeon 3210.

So, we supposes Svwanpipe-i386-5.10.pkg or Svzaptel-i386-5.10.pkg from
Sangoma checked up a little and on some one processor and doesn't heave
up even on beta, as they are declared, at the best on alpha


What's the reason of error???

      * wanpipe doesn't work under Solaris?
        
      * wrong working signalling with zaptel? (but same configuration
        works good under Linux)
        
      * wrong working of FreeSWITCH, which was built correctly, but
        their work was violated?
      * 
      * 
Thanks, Evgeniy.






From msc at freeswitch.org  Fri Dec  5 07:08:18 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 07:08:18 -0800
Subject: [Freeswitch-users] Handling directory of sound files
Message-ID: <B25E3A72-98AE-4013-AAE0-CE978FD28717@freeswitch.org>

Check out mod_localstream on the wiki and see if that sounds like what  
you need. I'm still learning it all myself but I believe that's where  
you should start. Please report back with any questions and we will  
take it from there!

-MC


On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>  
wrote:

> Its not without music on hold completely. Say, e.g, moh is being  
> played but when i press 1 it should start playing files contained in  
> a specific directory sequentially or randomly. I havent got any  
> solution to this problem yet. Can anyone plz guide me to some  
> documentation or anything else regarding this matter.
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>     
>                                                                      
> Faisal
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frank at impactfax.com  Fri Dec  5 07:22:13 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 10:22:13 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <20804652.post@talk.nabble.com>
Message-ID: <1fdd01c956ed$4003d280$33014c0a@ws4>


Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone (a key press, say 2) and when FS
hears that tone, then FS can broadcast another key tone (say 6) back to
the channels?





From pmhshz at gmail.com  Fri Dec  5 07:31:01 2008
From: pmhshz at gmail.com (shehzad p)
Date: Fri, 5 Dec 2008 07:31:01 -0800 (PST)
Subject: [Freeswitch-users] How to setup TLS between two Freeswitch servers
Message-ID: <20856369.post@talk.nabble.com>



I am wondering how to setup two freeswitch servers to route call with TLS
configured between them.

As shown in wiki http://wiki.freeswitch.org/wiki/SIP_TLS, I created two
certificates on one freeswitch, and changed SIP profile by enabling tls in
it,
then Starting freeswitch it just opens port 5061 (for TLS ),

But when i route the call from that FS server, it uses the its general ports
(5060 and 5080) for call.

Where i am missing something?, 
A doubt is about where to place which certificate. 
-- 
View this message in context: http://www.nabble.com/How-to-setup-TLS-between-two-Freeswitch-servers-tp20856369p20856369.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From carole.olivier at enst.fr  Fri Dec  5 07:35:51 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Fri, 5 Dec 2008 07:35:51 -0800 (PST)
Subject: [Freeswitch-users] conference configured to call automatically the
 attended does not work
Message-ID: <20856465.post@talk.nabble.com>


Hello,

I have got some problems for the configuration of a simple conference which
should be established by calling an extension and automatically inviting 2
people.

Actually, this is based on the default configuration of Freeswitch
(extension 0911). I have changed it a little:

<extension name="conference_test">
    <condition field="destination_number" expression="^0911$">
    <!--  These parameters effect the outcalls made once you join 
      --> 
      <action application="set"
data="conference_auto_outcall_caller_id_name=telephoneX" /> 
      <action application="set"
data="conference_auto_outcall_id_number=0911" /> 
      <action application="set" data="conference_auto_outcall_timeout=60" /> 
      <action application="set" data="conference_auto_outcall_flags=none" /> 
 
    <!--  list of devices called  --> 
      <action application="conference_set_auto_outcall"
data="sofia/internal/1010@$${domain}" /> 
      <action application="conference_set_auto_outcall"
data="sofia/internal/1002@$${domain}" /> 

      <action application="conference" data="conferenceTest at default" /> 
    </condition>
 </extension>


I have attached a file with the console errors. There are some errors (moh
errors) but since these were also present for room conference and it did not
prevent it for working, I guess this is not the fundamental reason for the
previous problem.

I have an additional question. I have installed freeswitch from
opensuse.org, there is a simple "one-click installation" but I am not sure
this was a good idea, it seems to be light isn't?

Thanks for your help,
Carole
http://www.nabble.com/file/p20856465/error_console.txt error_console.txt 
-- 
View this message in context: http://www.nabble.com/conference-configured-to-call-automatically-the-attended-does-not-work-tp20856465p20856465.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Fri Dec  5 07:36:19 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 09:36:19 -0600
Subject: [Freeswitch-users] How to setup TLS between two Freeswitch
	servers
In-Reply-To: <20856369.post@talk.nabble.com>
References: <20856369.post@talk.nabble.com>
Message-ID: <7D08D1D5-5FA8-40FD-BF82-EA9412F6E0D2@freeswitch.org>

You would use something like this sofia/profile/ 
user at remotefsip;transport=tls

/b

On Dec 5, 2008, at 9:31 AM, shehzad p wrote:

>
>
> I am wondering how to setup two freeswitch servers to route call  
> with TLS
> configured between them.




From msc at freeswitch.org  Fri Dec  5 07:44:30 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 07:44:30 -0800
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <1fdd01c956ed$4003d280$33014c0a@ws4>
References: <1fdd01c956ed$4003d280$33014c0a@ws4>
Message-ID: <C7F15A37-CCA9-4CAE-AA30-C012224BFBC6@freeswitch.org>

What would need to happen after the tone is sent back out? Also, would  
this be part of something like an IVR?

-MC


On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

>
> Is there any dialplan instructions that could be added that would sit
> and listen during a call for a tone (a key press, say 2) and when FS
> hears that tone, then FS can broadcast another key tone (say 6) back  
> to
> the channels?
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From frank at impactfax.com  Fri Dec  5 07:51:07 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 10:51:07 -0500
Subject: [Freeswitch-users] key tone trigger event during call
Message-ID: <201201c956f1$49abe940$33014c0a@ws4>

Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone (a key press, say 2) and when FS
hears that tone, then FS can broadcast another key tone (say 6) back to
the channels?
 
-Frank
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From msc at freeswitch.org  Fri Dec  5 07:56:32 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 07:56:32 -0800
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <49391D40.6050103@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
Message-ID: <70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>

I will do some research on this and let you know what I find out.  
Question: are these internal calls or pstn or ?? Just curious about  
your environment.

Thanks,
MC



On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>  
wrote:

> The proto_specific_hangup_cause is missing on one of the two
> call legs. When the caller hangs up it is missing from the a-leg CDR.
> When the callee hangs up it is missing from the b-leg CDR. Is this  
> nornal?
>
> And another question : what piece of info could inform me about who
> hanged up?
>
>
> -- 
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From frank at impactfax.com  Fri Dec  5 08:00:17 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 11:00:17 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <C7F15A37-CCA9-4CAE-AA30-C012224BFBC6@freeswitch.org>
Message-ID: <203101c956f2$91781450$33014c0a@ws4>

After the tone is sent back out, we are done.  There is nothing left to
do.  
No, this key press detection is during a bridged call between two
parties.  No IVR here.   So, FS hears a key press tone during a call and
then responds to the parties with another/different key press tone.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
 
What would need to happen after the tone is sent back out? Also, would  
this be part of something like an IVR?

-MC


On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

>
> Is there any dialplan instructions that could be added that would sit
> and listen during a call for a tone (a key press, say 2) and when FS
> hears that tone, then FS can broadcast another key tone (say 6) back  
> to
> the channels?
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From msc at freeswitch.org  Fri Dec  5 08:02:35 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 08:02:35 -0800
Subject: [Freeswitch-users] Event_Socket: How to determine the
	ogininater uuid before an outgoing call is set up?
In-Reply-To: <4939168E.9020400@gmx.net>
References: <4939168E.9020400@gmx.net>
Message-ID: <F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>

What is your originate string?
-MC


On Dec 5, 2008, at 3:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> I am building an IVR application where an incoming call is  
> initiating an
> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
> incoming channel) at originate time, I am able to reference to the
> incomig call, once the outgoing call is set up. So the outgoing call  
> can
> see the uuid of the incoming call who has originated the outgoing  
> call.
> This is needed for bridging the 2 calls together.
>
> However I want to control also the call setup process (see, if the
> outgoing call is ringing etc.). At call setup time, when I parse the
> channel_originate ,channel_outgoing and channel_progress events, I
> cannot see any reference to the incoming call (variable_other_uuid is
> not set). I suspect that variables are only passed once the outgoing
> channel is set up.
>
> Has anybody an idea, how I may get the uuid of the originating uuid in
> the outgoing call at call setup?
>
> Best regards
> Peter
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Fri Dec  5 08:07:42 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 10:07:42 -0600
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <203101c956f2$91781450$33014c0a@ws4>
References: <203101c956f2$91781450$33014c0a@ws4>
Message-ID: <CDF5931B-8C68-4D05-8D06-EA8095D8AC96@freeswitch.org>

So receive DTMF respond with more DTMF?

/b

On Dec 5, 2008, at 10:00 AM, Frank @ Impact wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.




From msc at freeswitch.org  Fri Dec  5 08:07:39 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 08:07:39 -0800
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
In-Reply-To: <1228488785.5481.4.camel@opos20.altron.lan>
References: <1228488785.5481.4.camel@opos20.altron.lan>
Message-ID: <579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>

Evgeniy,

I will need some time to digest all of this. I have an a104 but I  
don't have a solaris system for testing. I will report back as soon as  
I can.

-MC



On Dec 5, 2008, at 6:53 AM, Evgeniy Zolotov <zolotov at altron.ua> wrote:

>  Greetings!
>
> Question about possibility of the use FreeSWITCH for work with T1/E1
> streams under Sun Solaris 10 a bit clears up (Solaris 11 is in  
> condition
> of alpha-version and not suitable for the industrial use). But answers
> carry more negative sense.
>
> Start of T1/E1 under Sun Solaris has 2 stages: a) start of wanpipe's
> interface; b) make FreeSWITCH for Sun Solaris.
>
>
> a) For Linux Sangoma recommends
> (http://wiki.sangoma.com/wanpipe-freeswitch-install) installation of  
> new
> interface wanpipe-3.3.14.tgz, which has beta-status and placed at
> ftp://ftp.sangoma.com/linux/current_wanpipe.
>
> This installation and tests were successful.
>
> In this release (2007 y.) by Sangoma was added TDM API - native
> interface for FreeSwitch (YATE) ? which is absent in all previous
> releases of WANPIPE. In the Linux release 3.3.14 this variant was  
> named
> as ?TDM API?, in older releases - ?TDM Voice?.
>
> In this case FreeSWITCH works very good with E1/T1 cards
> A-101/102/104/108 without installation in zaptel system, and spans in
> configuration files of FreeSWITCH declared as [wanpipe#].
>
> But for Sun Solaris we found only
> drverftp://ftp.sangoma.com/Solaris/Beta/SVwanpipe-i386-5.10.pkgbeta,
> which dated 2007:
>
>> NAME=sangoma.com Wanpipe Driver
>
>> VERSION=1.1.0,REV=2007-07-16,
>
> Packet Svwanpipe-i386-5.10.pkg uses Svzaptel-i386-5.10.pkg, which  
> wasn't
> developed by Sangoma, but by little free
> communityhttp://www.solarisvoip.com/, source codes of this packet is
> herehttps://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk/
>
> The analysis of source codes shows that this project develops very
> slowly (there are no updates for about year), has very limited
> functionality and supports ( unlike original zaptel ) very limited  
> list
> of cards ( only one ;) - Digium Wildcard TE110P T1/PRI).
>
> Svwanpipe-i386-5.10.pkg supports only 64-bit Sun Solaris (on CD, which
> we get with Sangoma's cards, presents 32-bit driver and PDF document
> about installation under Sun Solaris ? but it dated 2001 ? 2002 yy 
> ).
>
> At first we have checked up installation with TDM Voice + zaptel (like
> for Asterisk) under Linux. We configured PRI spans as [zt] ... - such
> installation works good and we could do calls :
>
>
>> originate openzap/1/A/20000 &sleep(3)
>
>
>      * evidently that call retranslates from span 1 to span 2,
>        connected with cross-cable, and goes to extension 20000.
>
>
>
> b) About making FreeSwitch:
>
> Under Sun Solaris 10 with GCC makes FreeSWITCH core and most modules,
> except openzapand some others, because Sun Solaris 10 has GCC 3.4,
> but ./configure for openzap requires compatibility with ANSII 99.
>
> We have checked 3 different methods of makinf FreeSWITCH :
>
>
> 1. set up GCC 4.0.2 from CSW-repositaries (and all reguired for GCC
> *.pkg);
>
> 2. set up SunStudio 12 and do make cc/CC;
>
> 3.cross-compile FreeSwitch for Sun Solaris under Linux with CC- 
> options.
>
>
> In testing we used assembly SunStudio as 32-bit application (64-bit
> comes to the end with mistakes of assembly of some libraries, it is
> possible to correct for it easily, but we did not begin to specify  
> it).
>
>
> For testing cards A-104 we have repeated the same installation and
> configuring as under Linux (wanpipe TDM Voice + zaptel) on 4 different
> 64-bit servers under Sun Solaris 10:
>
>
>      * on 3 servers (2 of them manufactured by SUN) executed
>        successfully : # wanrouter start
>
>
>
> 4 wanrouter# interfaces was created; leds on spans, connected with
> cross-cable, becomes GREEN, i.e. synchronization T1/E1 presents (no
> alarms).
>
> On these servers FreeSWITCH correctly makes:
>
>
>> load mod_openzap
>
>
>      * but when we make :
>
>
>
>> originate openzap/1/A/20000 &sleep(3)
>
>
> - for 2 connected with crosss-cable spans (1 & 2) FreeSWITCH transmits
> PRI message (chan 1/31), but chan 2/31 doesn't receive this message
> (unlike under Linux) and call breaks after timeout.
>
> On 4-th server wanpipe doesn't even starts by mistake some IOCTL (i.e.
> at a command for device). Three ?working? servers is on AMD Opteron  
> Dual
> Core 2214(F), fourth is on ntel Xeon 3210.
>
> So, we supposes Svwanpipe-i386-5.10.pkg or Svzaptel-i386-5.10.pkg from
> Sangoma checked up a little and on some one processor and doesn't  
> heave
> up even on beta, as they are declared, at the best on alpha
>
>
> What's the reason of error???
>
>      * wanpipe doesn't work under Solaris?
>
>      * wrong working signalling with zaptel? (but same configuration
>        works good under Linux)
>
>      * wrong working of FreeSWITCH, which was built correctly, but
>        their work was violated?
>      *
>      *
> Thanks, Evgeniy.
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From frank at impactfax.com  Fri Dec  5 08:08:01 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 11:08:01 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <CDF5931B-8C68-4D05-8D06-EA8095D8AC96@freeswitch.org>
Message-ID: <205401c956f3$a60b4d50$33014c0a@ws4>

Yes.  listen in for 1 DTMF during a call and then signal back a
different DTMF.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
 
So receive DTMF respond with more DTMF?

/b

On Dec 5, 2008, at 10:00 AM, Frank @ Impact wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From Claudio.Cavalera at italtel.it  Fri Dec  5 08:09:12 2008
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Fri, 5 Dec 2008 17:09:12 +0100
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <203101c956f2$91781450$33014c0a@ws4>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0969622@BESONE.corp.dom>

freeswitch-users-bounces at lists.freeswitch.org wrote:
> After the tone is sent back out, we are done.  There is
> nothing left to
> do.


Maybe you can look at:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
Ciao,
Claudio


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From regs at kinetix.gr  Fri Dec  5 08:11:47 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 18:11:47 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
Message-ID: <493952C3.9060202@kinetix.gr>

Both legs are SIP. From non-registered endpoints (if of any use).

Michael S Collins wrote:
> I will do some research on this and let you know what I find out.  
> Question: are these internal calls or pstn or ?? Just curious about  
> your environment.
>
> Thanks,
> MC
>
>
>
> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>  
> wrote:
>
>   
>> The proto_specific_hangup_cause is missing on one of the two
>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>> When the callee hangs up it is missing from the b-leg CDR. Is this  
>> nornal?
>>
>> And another question : what piece of info could inform me about who
>> hanged up?
>>
>>
>> -- 
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From brian at freeswitch.org  Fri Dec  5 08:12:50 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 10:12:50 -0600
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
In-Reply-To: <579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>
References: <1228488785.5481.4.camel@opos20.altron.lan>
	<579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>
Message-ID: <43F6603A-909B-4B35-A04A-FE574814ECB9@freeswitch.org>

Does it list wanpipe TDM support on the Solaris builds of wanpipe?  I  
wasn't aware the TDM stuff was ported yet.

/b

On Dec 5, 2008, at 10:07 AM, Michael S Collins wrote:

> Evgeniy,
>
> I will need some time to digest all of this. I have an a104 but I
> don't have a solaris system for testing. I will report back as soon as
> I can.
>
> -MC




From regs at kinetix.gr  Fri Dec  5 08:17:07 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 18:17:07 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
Message-ID: <49395403.6080404@kinetix.gr>

I am sending this second email to request/suggest/enquire about 
something relevant :

Wouldn't it be useful to know which end of a specific call leg send the 
protocol
specific hangup cause? Otherwise it would be difficult to understand 
what really happened.



Michael S Collins wrote:
> I will do some research on this and let you know what I find out.  
> Question: are these internal calls or pstn or ?? Just curious about  
> your environment.
>
> Thanks,
> MC
>
>
>
> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>  
> wrote:
>
>   
>> The proto_specific_hangup_cause is missing on one of the two
>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>> When the callee hangs up it is missing from the b-leg CDR. Is this  
>> nornal?
>>
>> And another question : what piece of info could inform me about who
>> hanged up?
>>
>>
>> -- 
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

-------------- next part --------------
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From msc at freeswitch.org  Fri Dec  5 08:20:11 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 08:20:11 -0800
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <203101c956f2$91781450$33014c0a@ws4>
References: <203101c956f2$91781450$33014c0a@ws4>
Message-ID: <041F4629-11AD-4836-803F-9CD891454C3D@freeswitch.org>

Will the call be terminated at that point or does it need to continue?  
I do know that the tone_detect app can listen for a dtmf from either  
direction and can trigger execution of another app/extension/etc.  
However, I've never tried it on a bridged call, so I'm curious to see  
what would happen. The other question I would need to research is what  
would happen if the dtmf was sent rfc2833 style.

Hop on the wiki and look at tone_detect while I research the other  
questions and we will see what we can come up with.

-MC


On Dec 5, 2008, at 8:00 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-
>
> What would need to happen after the tone is sent back out? Also, would
> this be part of something like an IVR?
>
> -MC
>
>
> On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>
> wrote:
>
>>
>> Is there any dialplan instructions that could be added that would sit
>> and listen during a call for a tone (a key press, say 2) and when FS
>> hears that tone, then FS can broadcast another key tone (say 6) back
>> to
>> the channels?
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Fri Dec  5 08:22:53 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Dec 2008 11:22:53 -0500
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
In-Reply-To: <43F6603A-909B-4B35-A04A-FE574814ECB9@freeswitch.org>
References: <1228488785.5481.4.camel@opos20.altron.lan>
	<579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>
	<43F6603A-909B-4B35-A04A-FE574814ECB9@freeswitch.org>
Message-ID: <21906AF3-7B10-4097-88AC-48F245E0498A@jerris.com>

Last I spoke to doug at sangoma, solaris support is still not in their  
platform abstraction lib (there are drivers).  Please contact sangoma  
sales and request this.

Mike.

p.s. make sure to tell them its for FreeSWITCH


On Dec 5, 2008, at 11:12 AM, Brian West wrote:

> Does it list wanpipe TDM support on the Solaris builds of wanpipe?  I
> wasn't aware the TDM stuff was ported yet.
>
> /b
>
> On Dec 5, 2008, at 10:07 AM, Michael S Collins wrote:
>
>> Evgeniy,
>>
>> I will need some time to digest all of this. I have an a104 but I
>> don't have a solaris system for testing. I will report back as soon  
>> as
>> I can.
>>
>> -MC
>



From brian at freeswitch.org  Fri Dec  5 08:23:13 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 10:23:13 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493952C3.9060202@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<493952C3.9060202@kinetix.gr>
Message-ID: <1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>

Did you say what SVN rev you're running.

/b

On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:

> Both legs are SIP. From non-registered endpoints (if of any use).




From frank at impactfax.com  Fri Dec  5 08:29:57 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 11:29:57 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <041F4629-11AD-4836-803F-9CD891454C3D@freeswitch.org>
Message-ID: <207d01c956f6$b6165430$33014c0a@ws4>

The call should continue after FS hears the key press and responds with
its own key press tone.   Then the call just continues on.  They key
press from one of the parties would come some time after the call is
bridged.  Maybe some 10 or 20 seconds into the call for example.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins
 
Will the call be terminated at that point or does it need to continue?  
I do know that the tone_detect app can listen for a dtmf from either  
direction and can trigger execution of another app/extension/etc.  
However, I've never tried it on a bridged call, so I'm curious to see  
what would happen. The other question I would need to research is what  
would happen if the dtmf was sent rfc2833 style.

Hop on the wiki and look at tone_detect while I research the other  
questions and we will see what we can come up with.

-MC


On Dec 5, 2008, at 8:00 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-
>
> What would need to happen after the tone is sent back out? Also, would
> this be part of something like an IVR?
>
> -MC
>
>
> On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>
> wrote:
>
>>
>> Is there any dialplan instructions that could be added that would sit
>> and listen during a call for a tone (a key press, say 2) and when FS
>> hears that tone, then FS can broadcast another key tone (say 6) back
>> to
>> the channels?
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From frank at impactfax.com  Fri Dec  5 08:32:54 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 11:32:54 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <041F4629-11AD-4836-803F-9CD891454C3D@freeswitch.org>
Message-ID: <208801c956f7$1f80dee0$33014c0a@ws4>

Looks like tone detect might do it.  But..

If so, What frequency would we use for particular keys?  

Will tone_Detect sniff both legs or would we just do both r and w on the
called leg?

Can the tone_Detect timeout just be a very large number or can we leave
out the timeout value so there is no timeout?

Could the trigger from tone Detect do a gentone for a certain key?

Not much on the wiki on the mod.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins


Hop on the wiki and look at tone_detect while I research the other  
questions and we will see what we can come up with.

-MC


On Dec 5, 2008, at 8:00 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> After the tone is sent back out, we are done.  There is nothing left  
> to
> do.
> No, this key press detection is during a bridged call between two
> parties.  No IVR here.   So, FS hears a key press tone during a call  
> and
> then responds to the parties with another/different key press tone.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-
>
> What would need to happen after the tone is sent back out? Also, would
> this be part of something like an IVR?
>
> -MC
>
>
> On Dec 5, 2008, at 7:22 AM, "Frank @ Impact" <frank at impactfax.com>
> wrote:
>
>>
>> Is there any dialplan instructions that could be added that would sit
>> and listen during a call for a tone (a key press, say 2) and when FS
>> hears that tone, then FS can broadcast another key tone (say 6) back
>> to
>> the channels?
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From cstomi.levlist at gmail.com  Fri Dec  5 08:34:45 2008
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Fri, 05 Dec 2008 17:34:45 +0100
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
References: <20677406.post@talk.nabble.com>	<BA08ED43-200A-4037-935D-B15716DC09AD@jerris.com>	<191c3a030811250600n5ba54fc0qb219b09e19726adf@mail.gmail.com>	<20687620.post@talk.nabble.com>	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>	<20689832.post@talk.nabble.com>	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
Message-ID: <49395825.2010008@gmail.com>

Hello,

I have the same problem,

I don't understand the difference between

progress_timeout
originate_timeout
call_timeout.

I log timelimit_sec in switch_ivr_originate function and it seems,
if I set call_timeout then, timelimit_sec will be this value
if I set originate_timeout then timelimit_sec will be this value. maybe 
this is for backward compat?

originate_timeout as in the wiki:
 "Determines how long FreeSwitch is going to wait for a response from 
the invite message sent to the gateway. "

I quess this would be an 100 reply.

But how could I set a timeout for 200? I mean timeout for an answer.
which variable would control this?
I quess it was call_timeout previosly.
Please explain me this timeout variables

Thanks,
Tamas

Michael Collins ?rta:
> FYI, it is on the channel variables page but it's in a crazy place under
> "unknown functionality" which is silly.
> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
>
> Anyway, I've got wiki cleaning on my to-do list and I'll start in earnest
> next month when I have some time...
>
> -MC
>
> On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>   
>>> I used "call-timeout" because that's how I understood it from the Wiki.
>>>  (?)
>>>
>>>       
>> Yep, that's all that there is on the wiki. Unfortunately the channel
>> variables page is one of many in need of some attention. I will add
>> "originate_timeout" right away. The only question remaining is what, if
>> anything, does call_timeout do? That channel variable is in the source code
>> but I don't know exactly what it does.
>>
>> -MC
>>
>>
>>     
>>> --
>>> View this message in context:
>>> http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>     
>
>   
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From msc at freeswitch.org  Fri Dec  5 08:35:40 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 08:35:40 -0800
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <49395403.6080404@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<49395403.6080404@kinetix.gr>
Message-ID: <5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>

Makes sense. I will look into this.
-MC


On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>  
wrote:

> I am sending this second email to request/suggest/enquire about  
> something relevant :
>
> Wouldn't it be useful to know which end of a specific call leg send  
> the protocol
> specific hangup cause? Otherwise it would be difficult to understand  
> what really happened.
>
>
>
> Michael S Collins wrote:
>>
>> I will do some research on this and let you know what I find out.
>> Question: are these internal calls or pstn or ?? Just curious about
>> your environment.
>>
>> Thanks,
>> MC
>>
>>
>>
>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>> wrote:
>>
>>
>>> The proto_specific_hangup_cause is missing on one of the two
>>> call legs. When the caller hangs up it is missing from the a-leg  
>>> CDR.
>>> When the callee hangs up it is missing from the b-leg CDR. Is this
>>> nornal?
>>>
>>> And another question : what piece of info could inform me about who
>>> hanged up?
>>>
>>>
>>> -- 
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> -- 
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Fri Dec  5 08:37:26 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 10:37:26 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<493952C3.9060202@kinetix.gr>
	<1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
Message-ID: <191c3a030812050837n4374e96fya71588a028869dc5@mail.gmail.com>

It's easy enough to set the value on both legs try r10614
It was only set on the opposing leg before but since it's harmless to set it
on both i did it for you.


On Fri, Dec 5, 2008 at 10:23 AM, Brian West <brian at freeswitch.org> wrote:

> Did you say what SVN rev you're running.
>
> /b
>
> On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
>
> > Both legs are SIP. From non-registered endpoints (if of any use).
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

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From anthony.minessale at gmail.com  Fri Dec  5 08:41:16 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 10:41:16 -0600
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <49395825.2010008@gmail.com>
References: <20677406.post@talk.nabble.com>
	<BA08ED43-200A-4037-935D-B15716DC09AD@jerris.com>
	<191c3a030811250600n5ba54fc0qb219b09e19726adf@mail.gmail.com>
	<20687620.post@talk.nabble.com>
	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>
	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>
	<20689832.post@talk.nabble.com>
	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
	<49395825.2010008@gmail.com>
Message-ID: <191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>

call_timeout is only used if you are bridging 2 channels where one or both
of them is still unanswered.

what you want to use is originate_timeout and forget about call_timeout

you also have
leg_timeout and leg_progress_timeout both to be set in the {}
that do the timeout from the perspective of the new channel leg instead of
the caller leg.



On Fri, Dec 5, 2008 at 10:34 AM, Tamas Cseke <cstomi.levlist at gmail.com>wrote:

> Hello,
>
> I have the same problem,
>
> I don't understand the difference between
>
> progress_timeout
> originate_timeout
> call_timeout.
>
> I log timelimit_sec in switch_ivr_originate function and it seems,
> if I set call_timeout then, timelimit_sec will be this value
> if I set originate_timeout then timelimit_sec will be this value. maybe
> this is for backward compat?
>
> originate_timeout as in the wiki:
>  "Determines how long FreeSwitch is going to wait for a response from
> the invite message sent to the gateway. "
>
> I quess this would be an 100 reply.
>
> But how could I set a timeout for 200? I mean timeout for an answer.
> which variable would control this?
> I quess it was call_timeout previosly.
> Please explain me this timeout variables
>
> Thanks,
> Tamas
>
> Michael Collins ?rta:
> > FYI, it is on the channel variables page but it's in a crazy place under
> > "unknown functionality" which is silly.
> > http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
> >
> > Anyway, I've got wiki cleaning on my to-do list and I'll start in earnest
> > next month when I have some time...
> >
> > -MC
> >
> > On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> >>> I used "call-timeout" because that's how I understood it from the Wiki.
> >>>  (?)
> >>>
> >>>
> >> Yep, that's all that there is on the wiki. Unfortunately the channel
> >> variables page is one of many in need of some attention. I will add
> >> "originate_timeout" right away. The only question remaining is what, if
> >> anything, does call_timeout do? That channel variable is in the source
> code
> >> but I don't know exactly what it does.
> >>
> >> -MC
> >>
> >>
> >>
> >>> --
> >>> View this message in context:
> >>>
> http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.html
> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>>
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From regs at kinetix.gr  Fri Dec  5 08:42:50 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 18:42:50 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<493952C3.9060202@kinetix.gr>
	<1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
Message-ID: <49395A0A.7070103@kinetix.gr>

FreeSWITCH Version 1.0.trunk (10579)

Brian West wrote:
> Did you say what SVN rev you're running.
>
> /b
>
> On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
>
>   
>> Both legs are SIP. From non-registered endpoints (if of any use).
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From jbr at consiglia.dk  Fri Dec  5 08:43:44 2008
From: jbr at consiglia.dk (Jon Bruel)
Date: Fri, 5 Dec 2008 17:43:44 +0100
Subject: [Freeswitch-users] How to force FS using the hosts entry in
	outgoing gateway calls
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>

For the configuration of a gateway I need to use a specific proxy domain name before the server (Covergence SBC with a BroadWorks Application Server behind) accepts calls. The twist is that the right proxy name points the wrong IP-address (the voicemail server for the account). I have tried to overrule this by adding a host entry (Linux). When I ping to the domain name I get the right address (the one from the host table), but the FS uses the address from the DNS lookup, not the address from the host table. What can I do to force the FS using the entry from the host table? Thanks /Jon.

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From anthony.minessale at gmail.com  Fri Dec  5 08:53:53 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 10:53:53 -0600
Subject: [Freeswitch-users] How to force FS using the hosts entry in
	outgoing gateway calls
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>
Message-ID: <191c3a030812050853p5257eeefvb421ff045ad3ef9d@mail.gmail.com>

set proxy to be the correct hostname and set register-proxy param to be the
correct IP


On Fri, Dec 5, 2008 at 10:43 AM, Jon Bruel <jbr at consiglia.dk> wrote:

>  For the configuration of a gateway I need to use a specific proxy domain
> name before the server (Covergence SBC with a BroadWorks Application Server
> behind) accepts calls. The twist is that the right proxy name points the
> wrong IP-address (the voicemail server for the account). I have tried to
> overrule this by adding a host entry (Linux). When I ping to the domain name
> I get the right address (the one from the host table), but the FS uses the
> address from the DNS lookup, not the address from the host table. What can I
> do to force the FS using the entry from the host table? Thanks /Jon.
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From Prometheus001 at gmx.net  Fri Dec  5 09:08:46 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 05 Dec 2008 18:08:46 +0100
Subject: [Freeswitch-users] Event_Socket: How to determine
 the	ogininater uuid before an outgoing call is set up?
In-Reply-To: <F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>
References: <4939168E.9020400@gmx.net>
	<F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>
Message-ID: <4939601E.7070601@gmx.net>

I am a step further, When I set the cid-name then I can access the data
dring
channel_outgoing
channel_originate
channel_progress
channel_answer

However setting the caller_caller_id_number might be better.

This is the originate request:
<?xml version="1.0" ?>
<methodCall>
<methodName>freeswitch.api</methodName>
<params>
<param><value><string>bgapi</string></value></param>
<param><value><string>originate
{other_unique_id=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_name=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_number=000,ignore_early_media=true}user/1001 at siplocal.safecomm.ch
&amp;transfer(5002)</string></value></param>
</params>
</methodCall>

Answer:
<?xml version="1.0" encoding="UTF-8"?>.
<methodResponse>.
<params>.
<param><value><string>+OK Job-UUID: 0856d3ec-c2e9-11dd-85f2-75efbd1bca02
</string></value></param>.
</params>.
</methodResponse>.

By the way: The Job-UUID is different from the channel uuid, so it
cannot be used for my issue.

Best regards
Peter

Michael S Collins schrieb:
> What is your originate string?
> -MC
>
>
> On Dec 5, 2008, at 3:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>
>   
>> I am building an IVR application where an incoming call is  
>> initiating an
>> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
>> incoming channel) at originate time, I am able to reference to the
>> incomig call, once the outgoing call is set up. So the outgoing call  
>> can
>> see the uuid of the incoming call who has originated the outgoing  
>> call.
>> This is needed for bridging the 2 calls together.
>>
>> However I want to control also the call setup process (see, if the
>> outgoing call is ringing etc.). At call setup time, when I parse the
>> channel_originate ,channel_outgoing and channel_progress events, I
>> cannot see any reference to the incoming call (variable_other_uuid is
>> not set). I suspect that variables are only passed once the outgoing
>> channel is set up.
>>
>> Has anybody an idea, how I may get the uuid of the originating uuid in
>> the outgoing call at call setup?
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From msc at freeswitch.org  Fri Dec  5 09:38:32 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 09:38:32 -0800
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <208801c956f7$1f80dee0$33014c0a@ws4>
References: <041F4629-11AD-4836-803F-9CD891454C3D@freeswitch.org>
	<208801c956f7$1f80dee0$33014c0a@ws4>
Message-ID: <87f2f3b90812050938s120f801dy29a95d02f601f89a@mail.gmail.com>

On Fri, Dec 5, 2008 at 8:32 AM, Frank @ Impact <frank at impactfax.com> wrote:
> Looks like tone detect might do it.  But..
>
> If so, What frequency would we use for particular keys?
>

http://en.wikipedia.org/wiki/DTMF#Keypad


> Will tone_Detect sniff both legs or would we just do both r and w on the
> called leg?
>

Just do both r and w.

> Can the tone_Detect timeout just be a very large number or can we leave
> out the timeout value so there is no timeout?

I know you can set it to a large number; I've never tried a "forever"
tone_detect. I'll check it out.

>
> Could the trigger from tone Detect do a gentone for a certain key?
>

I don't believe so. This is where the bind_meta_app functionality is
more applicable. The dialplan isn't really the best place to handle
"events" like this. (Event socket would be better if you can swing
that, but I think maybe a workaround is doable with just the dialplan
and some creativity.)

> Not much on the wiki on the mod.

My bad. I'm working on it. :) In the meantime grab those dtmf
frequency values and set up a test extension in your dialplan. Put a
tone_detect app in that test ext. (You could have the ext set tone
detect, sleep 10 seconds, do info app then hangup.) Then call the test
extension, press a few keys, then wait for the info app to dump. If
you have the tone_detect set a chan variable when a dtmf is pressed
then you'll see it in your info app dump. BTW, MikeJ reminded me about
the start_dtmf/stop_dtmf apps:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf

Those might be necessary if your dtmf's are not already in-band.

Here's a sample extension you could try for testing, dialing 9990:

<extension name="dtmf_detect_test">
    <condition field="destination_number" expression="^9990$">
        <action application="answer"/>
        <action application="tone_detect" data="DTMF1 697,1209 rw
+10000 set DTMF1=true"/>  <!-- detect DTMF 1 key-->
        <action application="sleep" data="10000"/>
        <action application="info"/>
        <action application="hangup"/>
    </condition>
</extension>

Give that a try and at least see if you can detect the tones...
-MC



From msc at freeswitch.org  Fri Dec  5 09:50:37 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 09:50:37 -0800
Subject: [Freeswitch-users] Event_Socket: How to determine the
	ogininater uuid before an outgoing call is set up?
In-Reply-To: <4939601E.7070601@gmx.net>
References: <4939168E.9020400@gmx.net>
	<F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>
	<4939601E.7070601@gmx.net>
Message-ID: <87f2f3b90812050950r2cfe7966h6be1146aa828b5ad@mail.gmail.com>

Peter,

thanks, I will ruminate on this and get back with you as soon as I can.
-MC

On Fri, Dec 5, 2008 at 9:08 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> I am a step further, When I set the cid-name then I can access the data
> dring
> channel_outgoing
> channel_originate
> channel_progress
> channel_answer
>
> However setting the caller_caller_id_number might be better.
>
> This is the originate request:
> <?xml version="1.0" ?>
> <methodCall>
> <methodName>freeswitch.api</methodName>
> <params>
> <param><value><string>bgapi</string></value></param>
> <param><value><string>originate
> {other_unique_id=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_name=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_number=000,ignore_early_media=true}user/1001 at siplocal.safecomm.ch
> &amp;transfer(5002)</string></value></param>
> </params>
> </methodCall>
>
> Answer:
> <?xml version="1.0" encoding="UTF-8"?>.
> <methodResponse>.
> <params>.
> <param><value><string>+OK Job-UUID: 0856d3ec-c2e9-11dd-85f2-75efbd1bca02
> </string></value></param>.
> </params>.
> </methodResponse>.
>
> By the way: The Job-UUID is different from the channel uuid, so it
> cannot be used for my issue.
>
> Best regards
> Peter
>
> Michael S Collins schrieb:
>> What is your originate string?
>> -MC
>>
>>
>> On Dec 5, 2008, at 3:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>
>>> I am building an IVR application where an incoming call is
>>> initiating an
>>> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
>>> incoming channel) at originate time, I am able to reference to the
>>> incomig call, once the outgoing call is set up. So the outgoing call
>>> can
>>> see the uuid of the incoming call who has originated the outgoing
>>> call.
>>> This is needed for bridging the 2 calls together.
>>>
>>> However I want to control also the call setup process (see, if the
>>> outgoing call is ringing etc.). At call setup time, when I parse the
>>> channel_originate ,channel_outgoing and channel_progress events, I
>>> cannot see any reference to the incoming call (variable_other_uuid is
>>> not set). I suspect that variables are only passed once the outgoing
>>> channel is set up.
>>>
>>> Has anybody an idea, how I may get the uuid of the originating uuid in
>>> the outgoing call at call setup?
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Fri Dec  5 09:57:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 11:57:51 -0600
Subject: [Freeswitch-users] Event_Socket: How to determine the
	ogininater uuid before an outgoing call is set up?
In-Reply-To: <87f2f3b90812050950r2cfe7966h6be1146aa828b5ad@mail.gmail.com>
References: <4939168E.9020400@gmx.net>
	<F616794F-3892-4B77-A16E-D26C85340BE2@freeswitch.org>
	<4939601E.7070601@gmx.net>
	<87f2f3b90812050950r2cfe7966h6be1146aa828b5ad@mail.gmail.com>
Message-ID: <191c3a030812050957v43ccaf33r5ba100cdcbf4e5e7@mail.gmail.com>

job-uuid can be used to match the BACKGROUND_JOB event which will
have the output of the originate command in the body.

since you are using bgapi it goes asyncronous and must deliver the reply to
you via the event interface.


On Fri, Dec 5, 2008 at 11:50 AM, Michael Collins <msc at freeswitch.org> wrote:

> Peter,
>
> thanks, I will ruminate on this and get back with you as soon as I can.
> -MC
>
> On Fri, Dec 5, 2008 at 9:08 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> > I am a step further, When I set the cid-name then I can access the data
> > dring
> > channel_outgoing
> > channel_originate
> > channel_progress
> > channel_answer
> >
> > However setting the caller_caller_id_number might be better.
> >
> > This is the originate request:
> > <?xml version="1.0" ?>
> > <methodCall>
> > <methodName>freeswitch.api</methodName>
> > <params>
> > <param><value><string>bgapi</string></value></param>
> > <param><value><string>originate
> >
> {other_unique_id=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_name=ed525a3a-c2e8-11dd-85f2-75efbd1bca02,origination_caller_id_number=000,ignore_early_media=true}user/
> 1001 at siplocal.safecomm.ch
> > &amp;transfer(5002)</string></value></param>
> > </params>
> > </methodCall>
> >
> > Answer:
> > <?xml version="1.0" encoding="UTF-8"?>.
> > <methodResponse>.
> > <params>.
> > <param><value><string>+OK Job-UUID: 0856d3ec-c2e9-11dd-85f2-75efbd1bca02
> > </string></value></param>.
> > </params>.
> > </methodResponse>.
> >
> > By the way: The Job-UUID is different from the channel uuid, so it
> > cannot be used for my issue.
> >
> > Best regards
> > Peter
> >
> > Michael S Collins schrieb:
> >> What is your originate string?
> >> -MC
> >>
> >>
> >> On Dec 5, 2008, at 3:54 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> >>
> >>
> >>> I am building an IVR application where an incoming call is
> >>> initiating an
> >>> outgoing call. When I pass a "variable_other_uuid" (the uuid of the
> >>> incoming channel) at originate time, I am able to reference to the
> >>> incomig call, once the outgoing call is set up. So the outgoing call
> >>> can
> >>> see the uuid of the incoming call who has originated the outgoing
> >>> call.
> >>> This is needed for bridging the 2 calls together.
> >>>
> >>> However I want to control also the call setup process (see, if the
> >>> outgoing call is ringing etc.). At call setup time, when I parse the
> >>> channel_originate ,channel_outgoing and channel_progress events, I
> >>> cannot see any reference to the incoming call (variable_other_uuid is
> >>> not set). I suspect that variables are only passed once the outgoing
> >>> channel is set up.
> >>>
> >>> Has anybody an idea, how I may get the uuid of the originating uuid in
> >>> the outgoing call at call setup?
> >>>
> >>> Best regards
> >>> Peter
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mehdi.chaabouni at gmail.com  Fri Dec  5 10:23:24 2008
From: mehdi.chaabouni at gmail.com (mehdix)
Date: Fri, 5 Dec 2008 10:23:24 -0800 (PST)
Subject: [Freeswitch-users]  Provider: Junction Networks
Message-ID: <20859688.post@talk.nabble.com>


I've got a problem with configuring a SIP trunk from Junction Networks with
FS: it only works for a few minutes then the line is dropped.

I tried Unlimitel with no problem.

Any Ideas?

Thanks
-- 
View this message in context: http://www.nabble.com/Provider%3A-Junction-Networks-tp20859688p20859688.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From msc at freeswitch.org  Fri Dec  5 10:23:37 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 10:23:37 -0800
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>
References: <20677406.post@talk.nabble.com>
	<191c3a030811250600n5ba54fc0qb219b09e19726adf@mail.gmail.com>
	<20687620.post@talk.nabble.com>
	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>
	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>
	<20689832.post@talk.nabble.com>
	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
	<49395825.2010008@gmail.com>
	<191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>
Message-ID: <87f2f3b90812051023v3aebd168r2d92ae44531d93bb@mail.gmail.com>

On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> call_timeout is only used if you are bridging 2 channels where one or both
> of them is still unanswered.
>
> what you want to use is originate_timeout and forget about call_timeout
>
> you also have
> leg_timeout and leg_progress_timeout both to be set in the {}
> that do the timeout from the perspective of the new channel leg instead of
> the caller leg.
>

I will make sure that the wiki reflects these explanations properly.
-MC



From brian at freeswitch.org  Fri Dec  5 10:30:46 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 10:30:46 -0800
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <20859688.post@talk.nabble.com>
References: <20859688.post@talk.nabble.com>
Message-ID: <04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>

What is the hangup cause?

/b

On Dec 5, 2008, at 10:23 AM, mehdix wrote:

> Any Ideas?




From jan.kubr at gmail.com  Fri Dec  5 10:33:21 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Fri, 5 Dec 2008 19:33:21 +0100
Subject: [Freeswitch-users] DTMF from cell phones
Message-ID: <698401620812051033lf758838m733191df67143cea@mail.gmail.com>

> > no solution. I have a similar problem, when calling Freeswitch from my
> > cell phone (via a SIP provider), sometimes DTMF is not recognized
>> The important thing to note is that when using
>> a SIP softphone (X-Lite) I have never had this problem, DTMF is


> So i guess that using latest version with a few changes in your config
> should work unless there's any other issue related to your sip provider
> ( PSTN / Media Gateway ), on this case you can get some captures of
> sip/rtp traffic to check SDP and rtp Marks.

I tried trunk and the values for the variables (all except
rtp-timer-name=none are already default in trunk), but only two things
are different:
1. When I press a key, the read app seem to always terminate, but not
always the dtmf is captured in a variable.
2. The read app seems to ignore the variable name parameter: calling
it with "1 1 104.wav choice_181152 10000 #" doesn't put the digit in
variable_choice_181152, but to dmtf_digit. Why is that?

> If it is coming from the sip provider as rfc 2833 dtmf, they are
> probably doing inband detection and failing at it.  If you look at an
> rtp dump you can confirm this.  If this is the case, there is nothing
> you can do on the FreeSWITCH side and the provider will have to fix it.

But the call goes through the same SIP provider even when using the
soft phone and there it works fine. The difference might be that then
it is SIP to SIP within the same provider.. How do I do the RTP dump?

Also I should have mentioned that DTMF is not captured only DURING the
file is being played. It is always captured correctly when I wait
until the playback is finished. Does this sound familiar? I thought
this would be somet obvious misconfiguration on my side.


Jan



From zolotov at altron.ua  Fri Dec  5 10:36:02 2008
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Fri, 05 Dec 2008 20:36:02 +0200
Subject: [Freeswitch-users] Troubles with FreeSWITCH and Sangoma	A104
In-Reply-To: <21906AF3-7B10-4097-88AC-48F245E0498A@jerris.com>
References: <1228488785.5481.4.camel@opos20.altron.lan>
	<579D2E32-459C-445B-AF64-B0B61C815753@freeswitch.org>
	<43F6603A-909B-4B35-A04A-FE574814ECB9@freeswitch.org>
	<21906AF3-7B10-4097-88AC-48F245E0498A@jerris.com>
Message-ID: <1228502162.5481.29.camel@opos20.altron.lan>

  Thanks to all for their answers.

1. to Michael Collins

> >> I will need some time to digest all of this. I have an a104 but I
> >> don't have a solaris system for testing. I will report back as soon  
> >> as I can

 We with impatience will wait for results of your tests. If there will
be any questions - we with pleasure will help you.

2. to Brian West

> > Does it list wanpipe TDM support on the Solaris builds of wanpipe?  I
> > wasn't aware the TDM stuff was ported yet.

There are 2 kind of TDM support into Sangoma. Into Linux installation
menu (ncurses) they named :
  #2 "TDM Voice", where signalling carries out with zaptel and
  #8 "TDM API"(libsangoma), where signalling carries out without zaptel,
this is native interface, which is used by FreeSWITCH and Yate.

In Svwanpipe-i386-5.10.pkg (for Sun Solaris) present "TDM Voice", but 
absent "TDM API". Despite the fact that "TDM Voice" is present, seems
it works incorrectly (it works good under Linux, but not Sun Solaris).

3. to Michael Jerris

> Last I spoke to doug at sangoma, solaris support is still not in their  
> platform abstraction lib (there are drivers).  Please contact sangoma  
> sales and request this.
> 
> Mike.
> 
> p.s. make sure to tell them its for FreeSWITCH

Michael, I sent the same messages to Sangona, but they similar ignore
them, because I have not received any answer from them.




? ???, 05/12/2008 ? 11:22 -0500, Michael Jerris ?????:
> Last I spoke to doug at sangoma, solaris support is still not in their  
> platform abstraction lib (there are drivers).  Please contact sangoma  
> sales and request this.
> 
> Mike.
> 
> p.s. make sure to tell them its for FreeSWITCH
> 
> 
> On Dec 5, 2008, at 11:12 AM, Brian West wrote:
> 
> > Does it list wanpipe TDM support on the Solaris builds of wanpipe?  I
> > wasn't aware the TDM stuff was ported yet.
> >
> > /b
> >
> > On Dec 5, 2008, at 10:07 AM, Michael S Collins wrote:
> >
> >> Evgeniy,
> >>
> >> I will need some time to digest all of this. I have an a104 but I
> >> don't have a solaris system for testing. I will report back as soon  
> >> as
> >> I can.
> >>
> >> -MC
> >
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gservat at gmail.com  Fri Dec  5 10:38:14 2008
From: gservat at gmail.com (Gonzalo Servat)
Date: Fri, 5 Dec 2008 16:38:14 -0200
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <87f2f3b90812051023v3aebd168r2d92ae44531d93bb@mail.gmail.com>
References: <20677406.post@talk.nabble.com> <20687620.post@talk.nabble.com>
	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>
	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>
	<20689832.post@talk.nabble.com>
	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
	<49395825.2010008@gmail.com>
	<191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>
	<87f2f3b90812051023v3aebd168r2d92ae44531d93bb@mail.gmail.com>
Message-ID: <dcc007e10812051038j7d2ef8eaxde73462c2ba4b1c4@mail.gmail.com>

On Fri, Dec 5, 2008 at 4:23 PM, Michael Collins <msc at freeswitch.org> wrote:

> On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > call_timeout is only used if you are bridging 2 channels where one or
> both
> > of them is still unanswered.
> >
> > what you want to use is originate_timeout and forget about call_timeout
> >
> > you also have
> > leg_timeout and leg_progress_timeout both to be set in the {}
> > that do the timeout from the perspective of the new channel leg instead
> of
> > the caller leg.
> >
>
> I will make sure that the wiki reflects these explanations properly.
>

Excellent :) I'm still wondering not 100% clear on the exact difference(s)
between call_timeout and originate_timeout ...

- Gonzalo
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From regs at kinetix.gr  Fri Dec  5 10:43:52 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Fri, 05 Dec 2008 20:43:52 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <191c3a030812050837n4374e96fya71588a028869dc5@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<493952C3.9060202@kinetix.gr>	<1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
	<191c3a030812050837n4374e96fya71588a028869dc5@mail.gmail.com>
Message-ID: <49397668.80808@kinetix.gr>

I tested it and it works fine but it got me thinking...

Is just a copy of the cause to the other leg the correct way
to do it? Couldn't the two call legs hang up with different causes?
Especially when I could override the cause before it got send
to the e.g. calling side using e.g. the hangup command?

To make myself clear : I could have the b-leg (in a bridge hangup)
sending me a user busy code and I could send a circuit/channel
unavailable to my caller (a-leg), let's
say because I don't trust my terminator (b-leg) and his codes and I want to
enforce another one and send it to my originator so that he could retry 
another
carrier.

What do you think?


Anthony Minessale wrote:
> It's easy enough to set the value on both legs try r10614
> It was only set on the opposing leg before but since it's harmless to 
> set it on both i did it for you.
>
>
> On Fri, Dec 5, 2008 at 10:23 AM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     Did you say what SVN rev you're running.
>
>     /b
>
>     On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
>
>     > Both legs are SIP. From non-registered endpoints (if of any use).
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From anthony.minessale at gmail.com  Fri Dec  5 10:46:05 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 12:46:05 -0600
Subject: [Freeswitch-users] Channel variable 'call_timeout'.
In-Reply-To: <dcc007e10812051038j7d2ef8eaxde73462c2ba4b1c4@mail.gmail.com>
References: <20677406.post@talk.nabble.com>
	<191c3a030811251144w55b5e3dgffd50e1005bd6cbc@mail.gmail.com>
	<87f2f3b90811251208i44f9234h3cdf2cd9fb7913f@mail.gmail.com>
	<20689832.post@talk.nabble.com>
	<87f2f3b90811251332s178e2b4g514cdf7c294f33af@mail.gmail.com>
	<87f2f3b90811251334x6cc2eac6o48569cbbc594220e@mail.gmail.com>
	<49395825.2010008@gmail.com>
	<191c3a030812050841i3a3e0b62s6610d08d5f8cff93@mail.gmail.com>
	<87f2f3b90812051023v3aebd168r2d92ae44531d93bb@mail.gmail.com>
	<dcc007e10812051038j7d2ef8eaxde73462c2ba4b1c4@mail.gmail.com>
Message-ID: <191c3a030812051046t5557da06w67d2cf3c2f09d657@mail.gmail.com>

forget call_timout it's your best bet

it's depricated.


On Fri, Dec 5, 2008 at 12:38 PM, Gonzalo Servat <gservat at gmail.com> wrote:

> On Fri, Dec 5, 2008 at 4:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > call_timeout is only used if you are bridging 2 channels where one or
>> both
>> > of them is still unanswered.
>> >
>> > what you want to use is originate_timeout and forget about call_timeout
>> >
>> > you also have
>> > leg_timeout and leg_progress_timeout both to be set in the {}
>> > that do the timeout from the perspective of the new channel leg instead
>> of
>> > the caller leg.
>> >
>>
>> I will make sure that the wiki reflects these explanations properly.
>>
>
> Excellent :) I'm still wondering not 100% clear on the exact difference(s)
> between call_timeout and originate_timeout ...
>
> - Gonzalo
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Dec  5 10:53:00 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 12:53:00 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <49397668.80808@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<493952C3.9060202@kinetix.gr>
	<1FA8A986-C608-4DEF-828F-E5189C464DB6@freeswitch.org>
	<191c3a030812050837n4374e96fya71588a028869dc5@mail.gmail.com>
	<49397668.80808@kinetix.gr>
Message-ID: <191c3a030812051053g266828c8w4de90e8abdd816e0@mail.gmail.com>

This variable is to specifically document the protocol specific last status
cause.
So you can know what the status was when you got a BYE or final response to
invite in the case of sip.
That's all it's for.





On Fri, Dec 5, 2008 at 12:43 PM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  I tested it and it works fine but it got me thinking...
>
> Is just a copy of the cause to the other leg the correct way
> to do it? Couldn't the two call legs hang up with different causes?
> Especially when I could override the cause before it got send
> to the e.g. calling side using e.g. the hangup command?
>
> To make myself clear : I could have the b-leg (in a bridge hangup)
> sending me a user busy code and I could send a circuit/channel
> unavailable to my caller (a-leg), let's
> say because I don't trust my terminator (b-leg) and his codes and I want to
> enforce another one and send it to my originator so that he could retry
> another
> carrier.
>
> What do you think?
>
>
> Anthony Minessale wrote:
>
> It's easy enough to set the value on both legs try r10614
> It was only set on the opposing leg before but since it's harmless to set
> it on both i did it for you.
>
>
> On Fri, Dec 5, 2008 at 10:23 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Did you say what SVN rev you're running.
>>
>> /b
>>
>> On Dec 5, 2008, at 10:11 AM, Apostolos Pantsiopoulos wrote:
>>
>> > Both legs are SIP. From non-registered endpoints (if of any use).
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From mehdi.chaabouni at gmail.com  Fri Dec  5 10:59:35 2008
From: mehdi.chaabouni at gmail.com (MEHDi CHAABOUNi)
Date: Fri, 5 Dec 2008 13:59:35 -0500
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
Message-ID: <c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>

Actually, i did not mean that the line is dropped during a call...
FS is configured to accept calls from the Junction Networks SIP trunk to
make an audio conference.
When I start FS and I dial the number all is working fine. But, if I wait
for a couple of minutes and then make my call I get an error recorded
message saying that the number is not in service...




On Fri, Dec 5, 2008 at 1:30 PM, Brian West <brian at freeswitch.org> wrote:

> What is the hangup cause?
>
> /b
>
> On Dec 5, 2008, at 10:23 AM, mehdix wrote:
>
> > Any Ideas?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Dec  5 11:05:14 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Dec 2008 11:05:14 -0800
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
Message-ID: <6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>

But you don't see the invite hitting FreeSWITCH?  And you're behind  
NAT?  Make it register every 30 seconds instead of the default 3600

/b

On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:

> Actually, i did not mean that the line is dropped during a call...
> FS is configured to accept calls from the Junction Networks SIP  
> trunk to make an audio conference.
> When I start FS and I dial the number all is working fine. But, if I  
> wait for a couple of minutes and then make my call I get an error  
> recorded message saying that the number is not in service...




From msc at freeswitch.org  Fri Dec  5 11:08:21 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 11:08:21 -0800
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
Message-ID: <87f2f3b90812051108v1ebb8809u25364323d9dfa6ad@mail.gmail.com>

Can you hit F8 and capture the debug output when making a call?
That'll help us see what's going on.
-MC

On Fri, Dec 5, 2008 at 10:59 AM, MEHDi CHAABOUNi
<mehdi.chaabouni at gmail.com> wrote:
> Actually, i did not mean that the line is dropped during a call...
> FS is configured to accept calls from the Junction Networks SIP trunk to
> make an audio conference.
> When I start FS and I dial the number all is working fine. But, if I wait
> for a couple of minutes and then make my call I get an error recorded
> message saying that the number is not in service...
>
>
>
>
> On Fri, Dec 5, 2008 at 1:30 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> What is the hangup cause?
>>
>> /b
>>
>> On Dec 5, 2008, at 10:23 AM, mehdix wrote:
>>
>> > Any Ideas?
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Dec  5 11:08:56 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Dec 2008 11:08:56 -0800
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
	<6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
Message-ID: <87f2f3b90812051108kcd34b5er4fc99e1f7e310aa2@mail.gmail.com>

Doh! Brian is way ahead of me, as usual...

On Fri, Dec 5, 2008 at 11:05 AM, Brian West <brian at freeswitch.org> wrote:
> But you don't see the invite hitting FreeSWITCH?  And you're behind
> NAT?  Make it register every 30 seconds instead of the default 3600
>
> /b
>
> On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:
>
>> Actually, i did not mean that the line is dropped during a call...
>> FS is configured to accept calls from the Junction Networks SIP
>> trunk to make an audio conference.
>> When I start FS and I dial the number all is working fine. But, if I
>> wait for a couple of minutes and then make my call I get an error
>> recorded message saying that the number is not in service...
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mehdi.chaabouni at gmail.com  Fri Dec  5 11:27:59 2008
From: mehdi.chaabouni at gmail.com (MEHDi CHAABOUNi)
Date: Fri, 5 Dec 2008 14:27:59 -0500
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
	<6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
Message-ID: <c9080b60812051127y56195007lad0ecfa3276c0dd1@mail.gmail.com>

I changed the parameter expire-seconds to 30. Now, I'm starting to see the
register request in the console.
I'll wait a couple of hours and get back to you guys.

Thanks

On Fri, Dec 5, 2008 at 2:05 PM, Brian West <brian at freeswitch.org> wrote:

> But you don't see the invite hitting FreeSWITCH?  And you're behind
> NAT?  Make it register every 30 seconds instead of the default 3600
>
> /b
>
> On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:
>
> > Actually, i did not mean that the line is dropped during a call...
> > FS is configured to accept calls from the Junction Networks SIP
> > trunk to make an audio conference.
> > When I start FS and I dial the number all is working fine. But, if I
> > wait for a couple of minutes and then make my call I get an error
> > recorded message saying that the number is not in service...
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jbr at consiglia.dk  Fri Dec  5 11:51:57 2008
From: jbr at consiglia.dk (Jon Bruel)
Date: Fri, 5 Dec 2008 20:51:57 +0100
Subject: [Freeswitch-users] How to force FS using the hosts entry in
	outgoing gateway calls
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A81417D@SBS2008SERVER.consiglia.local>

Thanks Anthony. Using the parameters: 

<param name="proxy" value="domain name"/>
<param name="register-proxy" value="ip address"/>

Returns error 900, and a 'ngrep port port-number' indicates that its doesn't try to register at all. I have now let the server look at a local DNS where I have added a "wrong" A-record. That solves the issue, but your solution would be cleaner. The version is: trunk 10220. /Jon




From anthony.minessale at gmail.com  Fri Dec  5 11:57:29 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2008 13:57:29 -0600
Subject: [Freeswitch-users] How to force FS using the hosts entry in
	outgoing gateway calls
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A81417D@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A81417D@SBS2008SERVER.consiglia.local>
Message-ID: <191c3a030812051157q2116c468j49516416db4978f7@mail.gmail.com>

you have an older revision.....

put sip:<ip>  instead of just <ip>

I recommend you update and either will work.


On Fri, Dec 5, 2008 at 1:51 PM, Jon Bruel <jbr at consiglia.dk> wrote:

> Thanks Anthony. Using the parameters:
>
> <param name="proxy" value="domain name"/>
> <param name="register-proxy" value="ip address"/>
>
> Returns error 900, and a 'ngrep port port-number' indicates that its
> doesn't try to register at all. I have now let the server look at a local
> DNS where I have added a "wrong" A-record. That solves the issue, but your
> solution would be cleaner. The version is: trunk 10220. /Jon
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From frank at impactfax.com  Fri Dec  5 17:09:00 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 20:09:00 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <87f2f3b90812050938s120f801dy29a95d02f601f89a@mail.gmail.com>
Message-ID: <25a201c9573f$3901df70$33014c0a@ws4>

I tried your suggested test.  Here is the business end of the extension
I tried.

        <action application="set" data="DTMF1=false"/>
        <action application="start_dtmf" data="true"/>
        <action application="tone_detect" data="DTMF1 697,1209 rw +10000
set DTMF1=true"/>  <!-- detect DTMF 1 key-->
        <action application="answer"/>
        <action application="sleep" data="10000"/>
        <action application="info"/>
        <action application="hangup"/>

but I always got DTMF1=false in the info dump.
I am using FS 9210

I have tried sending a call from my sip phone connected to an asterisk
server to FS (dial FS).  I also tried a PSTN call coming in on a PRI to
asterisk and then sip over to FS (another dial from asterisk).  In each
case, pressed 1 several times and the tone_detect never triggered.

Ideas?  Am I doing something stupid or is tone_detect not just right
here?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
 
Those might be necessary if your dtmf's are not already in-band.

Here's a sample extension you could try for testing, dialing 9990:

<extension name="dtmf_detect_test">
    <condition field="destination_number" expression="^9990$">
        <action application="answer"/>
        <action application="tone_detect" data="DTMF1 697,1209 rw
+10000 set DTMF1=true"/>  <!-- detect DTMF 1 key-->
        <action application="sleep" data="10000"/>
        <action application="info"/>
        <action application="hangup"/>
    </condition>
</extension>

Give that a try and at least see if you can detect the tones...
-MC

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From msc at freeswitch.org  Fri Dec  5 17:42:27 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 5 Dec 2008 17:42:27 -0800
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <25a201c9573f$3901df70$33014c0a@ws4>
References: <25a201c9573f$3901df70$33014c0a@ws4>
Message-ID: <5EC8A57C-5D9A-437C-9A7E-B87BAF4B752F@freeswitch.org>

That's a pretty old rev. Any chance you could make current?

-MC

Sent from my iPhone

On Dec 5, 2008, at 5:09 PM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> I tried your suggested test.  Here is the business end of the  
> extension
> I tried.
>
>        <action application="set" data="DTMF1=false"/>
>        <action application="start_dtmf" data="true"/>
>        <action application="tone_detect" data="DTMF1 697,1209 rw  
> +10000
> set DTMF1=true"/>  <!-- detect DTMF 1 key-->
>        <action application="answer"/>
>        <action application="sleep" data="10000"/>
>        <action application="info"/>
>        <action application="hangup"/>
>
> but I always got DTMF1=false in the info dump.
> I am using FS 9210
>
> I have tried sending a call from my sip phone connected to an asterisk
> server to FS (dial FS).  I also tried a PSTN call coming in on a PRI  
> to
> asterisk and then sip over to FS (another dial from asterisk).  In  
> each
> case, pressed 1 several times and the tone_detect never triggered.
>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
>
> Those might be necessary if your dtmf's are not already in-band.
>
> Here's a sample extension you could try for testing, dialing 9990:
>
> <extension name="dtmf_detect_test">
>    <condition field="destination_number" expression="^9990$">
>        <action application="answer"/>
>        <action application="tone_detect" data="DTMF1 697,1209 rw
> +10000 set DTMF1=true"/>  <!-- detect DTMF 1 key-->
>        <action application="sleep" data="10000"/>
>        <action application="info"/>
>        <action application="hangup"/>
>    </condition>
> </extension>
>
> Give that a try and at least see if you can detect the tones...
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gilbertandrew at me.com  Fri Dec  5 18:15:08 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Fri, 05 Dec 2008 21:15:08 -0500
Subject: [Freeswitch-users] How to force FS using the hosts entry
	in	outgoing gateway calls
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE4A814179@SBS2008SERVER.consiglia.local>
Message-ID: <9E3B7FE2-C459-40F4-A8B5-DBB064564F32@me.com>

Jon,

You should also be able to do a

'order hosts,bind' in /etc/hosts, no????


On Dec 5, 2008, at 11:43 AM, Jon Bruel wrote:

> For the configuration of a gateway I need to use a specific proxy  
> domain name before the server (Covergence SBC with a BroadWorks  
> Application Server behind) accepts calls. The twist is that the  
> right proxy name points the wrong IP-address (the voicemail server  
> for the account). I have tried to overrule this by adding a host  
> entry (Linux). When I ping to the domain name I get the right  
> address (the one from the host table), but the FS uses the address  
> from the DNS lookup, not the address from the host table. What can I  
> do to force the FS using the entry from the host table? Thanks /Jon.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mehdi.chaabouni at gmail.com  Fri Dec  5 18:34:22 2008
From: mehdi.chaabouni at gmail.com (MEHDi CHAABOUNi)
Date: Fri, 5 Dec 2008 21:34:22 -0500
Subject: [Freeswitch-users] Provider: Junction Networks
In-Reply-To: <c9080b60812051127y56195007lad0ecfa3276c0dd1@mail.gmail.com>
References: <20859688.post@talk.nabble.com>
	<04F6DE03-0254-49C2-B9FE-9F5DD215D0EE@freeswitch.org>
	<c9080b60812051059q367609fck27f006b5ca810d6e@mail.gmail.com>
	<6DC35159-22AB-431E-81C8-9A29DC0E6E9B@freeswitch.org>
	<c9080b60812051127y56195007lad0ecfa3276c0dd1@mail.gmail.com>
Message-ID: <c9080b60812051834w2cfd78d7ob81fa53b70563181@mail.gmail.com>

It's working... thanks a lot

On Fri, Dec 5, 2008 at 2:27 PM, MEHDi CHAABOUNi
<mehdi.chaabouni at gmail.com>wrote:

> I changed the parameter expire-seconds to 30. Now, I'm starting to see the
> register request in the console.
> I'll wait a couple of hours and get back to you guys.
>
> Thanks
>
>
> On Fri, Dec 5, 2008 at 2:05 PM, Brian West <brian at freeswitch.org> wrote:
>
>> But you don't see the invite hitting FreeSWITCH?  And you're behind
>> NAT?  Make it register every 30 seconds instead of the default 3600
>>
>> /b
>>
>> On Dec 5, 2008, at 10:59 AM, MEHDi CHAABOUNi wrote:
>>
>> > Actually, i did not mean that the line is dropped during a call...
>> > FS is configured to accept calls from the Junction Networks SIP
>> > trunk to make an audio conference.
>> > When I start FS and I dial the number all is working fine. But, if I
>> > wait for a couple of minutes and then make my call I get an error
>> > recorded message saying that the number is not in service...
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From frank at impactfax.com  Fri Dec  5 19:01:32 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 5 Dec 2008 22:01:32 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <5EC8A57C-5D9A-437C-9A7E-B87BAF4B752F@freeswitch.org>
Message-ID: <006d01c9574e$f1a70410$33014c0a@ws4>

Also got it on 9579 as well.  

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins
Sent: Friday, December 05, 2008 8:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call

That's a pretty old rev. Any chance you could make current?

-MC

Sent from my iPhone

On Dec 5, 2008, at 5:09 PM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> I tried your suggested test.  Here is the business end of the  
> extension
> I tried.
>
>        <action application="set" data="DTMF1=false"/>
>        <action application="start_dtmf" data="true"/>
>        <action application="tone_detect" data="DTMF1 697,1209 rw  
> +10000
> set DTMF1=true"/>  <!-- detect DTMF 1 key-->
>        <action application="answer"/>
>        <action application="sleep" data="10000"/>
>        <action application="info"/>
>        <action application="hangup"/>
>
> but I always got DTMF1=false in the info dump.
> I am using FS 9210
>
> I have tried sending a call from my sip phone connected to an asterisk
> server to FS (dial FS).  I also tried a PSTN call coming in on a PRI  
> to
> asterisk and then sip over to FS (another dial from asterisk).  In  
> each
> case, pressed 1 several times and the tone_detect never triggered.
>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
>
> Those might be necessary if your dtmf's are not already in-band.
>
> Here's a sample extension you could try for testing, dialing 9990:
>
> <extension name="dtmf_detect_test">
>    <condition field="destination_number" expression="^9990$">
>        <action application="answer"/>
>        <action application="tone_detect" data="DTMF1 697,1209 rw
> +10000 set DTMF1=true"/>  <!-- detect DTMF 1 key-->
>        <action application="sleep" data="10000"/>
>        <action application="info"/>
>        <action application="hangup"/>
>    </condition>
> </extension>
>
> Give that a try and at least see if you can detect the tones...
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Fri Dec  5 22:01:25 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Dec 2008 00:01:25 -0600
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <25a201c9573f$3901df70$33014c0a@ws4>
References: <25a201c9573f$3901df70$33014c0a@ws4>
Message-ID: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>

make current or install current svn on a different box.

/b

On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:

>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?




From pmhshz at gmail.com  Fri Dec  5 22:42:52 2008
From: pmhshz at gmail.com (shehzad p)
Date: Fri, 5 Dec 2008 22:42:52 -0800 (PST)
Subject: [Freeswitch-users] How to setup TLS between two Freeswitch
 servers
In-Reply-To: <7D08D1D5-5FA8-40FD-BF82-EA9412F6E0D2@freeswitch.org>
References: <20856369.post@talk.nabble.com>
	<7D08D1D5-5FA8-40FD-BF82-EA9412F6E0D2@freeswitch.org>
Message-ID: <20867323.post@talk.nabble.com>


thanks Brian,

thank you so much for useful reply, It works very well now :)...

-
msp


Brian West-3 wrote:
> 
> You would use something like this sofia/profile/ 
> user at remotefsip;transport=tls
> 
> /b
> 
> On Dec 5, 2008, at 9:31 AM, shehzad p wrote:
> 
>>
>>
>> I am wondering how to setup two freeswitch servers to route call  
>> with TLS
>> configured between them.
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/How-to-setup-TLS-between-two-Freeswitch-servers-tp20856369p20867323.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From faisalmaqsoodi at yahoo.com  Fri Dec  5 23:42:43 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Fri, 5 Dec 2008 23:42:43 -0800 (PST)
Subject: [Freeswitch-users] Handling directory of sound files
In-Reply-To: <B25E3A72-98AE-4013-AAE0-CE978FD28717@freeswitch.org>
Message-ID: <427787.87769.qm@web30703.mail.mud.yahoo.com>

Thank u so much. mod local stream really works to play sound files from a local directory sequentially. Now can i jump to a specific file skipping the others? What should i use in dialplan? Have u any idea. Nothing is mentioned in the doc of mod lacal stream about that. 
?????????????????????????????????????????????????????? Faisal

--- On Fri, 12/5/08, Michael S Collins <msc at freeswitch.org> wrote:
From: Michael S Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Handling directory of sound files
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Date: Friday, December 5, 2008, 7:08 AM

Check out mod_localstream on the wiki and see if that sounds like what you need. I'm still learning it all myself but I believe that's where you should start. Please report back with any questions and we will take it from there!
-MC


On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi <faisalmaqsoodi at yahoo.com> wrote:

Its not without music on hold completely. Say, e.g, moh is being played
but when i press 1 it should start playing files contained in a
specific directory sequentially or randomly. I havent got any solution to this problem yet. Can anyone plz guide me to some documentation or anything else regarding this matter.
? ? ? ? ? ? ? ? ? ? ? ?? ???????????????????? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? Faisal


      _______________________________________________
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From faisalmaqsoodi at yahoo.com  Sat Dec  6 01:59:36 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sat, 6 Dec 2008 01:59:36 -0800 (PST)
Subject: [Freeswitch-users] Playing a file again and again
Message-ID: <202061.31652.qm@web30702.mail.mud.yahoo.com>

Hi,
??? Is there any built-in function, like playback, which plays a file again and again unless interrupted. I want to use a simple function not FIFO.
????????????????????????????????????????????????????????????????????????????? Faisal



      
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From odermann at googlemail.com  Sat Dec  6 02:13:01 2008
From: odermann at googlemail.com (Dennis)
Date: Sat, 6 Dec 2008 11:13:01 +0100
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <202061.31652.qm@web30702.mail.mud.yahoo.com>
References: <202061.31652.qm@web30702.mail.mud.yahoo.com>
Message-ID: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>

with sendmsg playback send: loops: -1

2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
> Hi,
>     Is there any built-in function, like playback, which plays a file again
> and again unless interrupted. I want to use a simple function not FIFO.
>
> Faisal
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From faisalmaqsoodi at yahoo.com  Sat Dec  6 06:50:55 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sat, 6 Dec 2008 06:50:55 -0800 (PST)
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>
Message-ID: <297073.11382.qm@web30701.mail.mud.yahoo.com>

I need some more help. I used send msg this way. Is there anything missing bcoz its not working. Plz let me know what else should i do. When i dial 2003, file is not played.
<extension name="Music">
sendmsg 2003
call-command: execute
execute-app-name: playback
execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav
loops: -1
? </extension>

--- On Sat, 12/6/08, Dennis <odermann at googlemail.com> wrote:
From: Dennis <odermann at googlemail.com>
Subject: Re: [Freeswitch-users] Playing a file again and again
To: freeswitch-users at lists.freeswitch.orgi
Date: Saturday, December 6, 2008, 2:13 AM

with sendmsg playback send: loops: -1

2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
> Hi,
>     Is there any built-in function, like playback, which plays a file
again
> and again unless interrupted. I want to use a simple function not FIFO.
>
> Faisal
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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From per_moeller at mac.com  Sat Dec  6 06:00:52 2008
From: per_moeller at mac.com (=?iso-8859-1?Q?Per_M=F8ller?=)
Date: Sat, 06 Dec 2008 15:00:52 +0100
Subject: [Freeswitch-users] Windows is slow?
In-Reply-To: <191c3a030812021205r619ad735le129731ccb8f69d0@mail.gmail.com>
References: <000001c9530d$912d86d0$b3889470$@com>
	<5800526b0812020813y5befb8f7p9ff6ca42cadb45b9@mail.gmail.com>
	<000f01c954b4$a616fa60$f244ef20$@com>
	<6E8D2069C08AA84A83D336E996AE4C6702336DC03C@mse17be1.mse17.exchange.ms>
	<191c3a030812021205r619ad735le129731ccb8f69d0@mail.gmail.com>
Message-ID: <002801c957ab$0e00b000$2a021000$@com>

No, only using a single local ip, no stun anywhere.

But I have found the time consuming application, it's DB.

When I commented out the following lines in the default dialplan, that a
call to a local extension would run through, there was no delay:

<action application="db"
data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
<action application="db"
data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_numb
er}"/>
<action application="db"
data="insert/${domain_name}-last_dial/global/${uuid}"/>
<anti-action application="db"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb
er}"/>
<anti-action application="db"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<anti-action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>

A cautious assumption would be that sqllite does not perform as well on
Win32. However I should mention it is compiled as debug, as I cannot get a
release version to compile.

// Per

Fra: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Anthony
Minessale
Sendt: 2. december 2008 21:05
Til: freeswitch-users at lists.freeswitch.org
Emne: Re: [Freeswitch-users] Windows is slow?

is it stun timeout ?

do you have one of the ip set to stun:foo ?

On Tue, Dec 2, 2008 at 1:33 PM, Michael Giagnocavo <mgg at giagnocavo.net>
wrote:
Can you do a console loglevel debug, then send all the output around that
time?

Apart from that, the quickest way might just to attach a debugger, then
break all when it pauses and see where the threads are :).

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Per
M?ller
Sent: Tuesday, December 02, 2008 12:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Windows is slow?

I checked out the trunk version, and it's still slow. However I found one
improvement - it does not crash on shutdown anymore.

Could anymore give me some pointers on how to try to debug this on the
Windows platform?


// Per

Fra: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Carlos
Talbot
Sendt: 2. december 2008 17:13
Til: freeswitch-users at lists.freeswitch.org
Emne: Re: [Freeswitch-users] Windows is slow?

Have you tried the latest msi build? It's based off svn 10564.

Carlos

On Sun, Nov 30, 2008 at 11:03 AM, Per M?ller <per_moeller at mac.com> wrote:
I have installed FS 1.0.0 on a Mac using the precompiled .dmg and FS 1.0.1
on a Windows Vista machine using the precompiled .msi - actually the same
machine).

Using the default configuration files, and using 2 Snom 360 phones I dialed
from extension 1000 to extension 1001. On the Mac, 1001 starts ringing
instantly, but under Windows it takes 1-2 seconds before it starts ringing.

It seems to be in the dialplan the time is spent. From the time I see this
line on the console:

[INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1000 in
context default

Until the next thing happens it always takes at least 1 full second, but on
the Mac it happens instantly.

Why is the Windows build this much slower? Is it a known problem?

I get the feeling that the majority of the FS community is Unix based, which
is fine by me, but I would really like to know just how well supported and
stable the Win32 build is and if this is currently a viable way to go, or if
I should stick to Linux/BSD/Mac for production use?


// Per


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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400




From odermann at googlemail.com  Sat Dec  6 08:13:17 2008
From: odermann at googlemail.com (Dennis)
Date: Sat, 6 Dec 2008 17:13:17 +0100
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <297073.11382.qm@web30701.mail.mud.yahoo.com>
References: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>
	<297073.11382.qm@web30701.mail.mud.yahoo.com>
Message-ID: <5e414ed0812060813k4877cc1anbae6a6fc2df9e4ba@mail.gmail.com>

although i do everything with socket outbound and php and not with the
xml-dialplans, i feel, that you are missing the basics.

first you should find out how to make a basic dialplan and how fs is
working. then you can start with playing soundfiles and then, how to
do a looped playback and other nice built in features.
there are lots of dialplan samples delivered with fs and the wiki will
help you to start with the rest.

dennis

2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
> I need some more help. I used send msg this way. Is there anything missing
> bcoz its not working. Plz let me know what else should i do. When i dial
> 2003, file is not played.
> <extension name="Music">
> sendmsg 2003
> call-command: execute
> execute-app-name: playback
> execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav
> loops: -1
>   </extension>
>
> --- On Sat, 12/6/08, Dennis <odermann at googlemail.com> wrote:
>
> From: Dennis <odermann at googlemail.com>
> Subject: Re: [Freeswitch-users] Playing a file again and again
> To: freeswitch-users at lists.freeswitch.orgi
> Date: Saturday, December 6, 2008, 2:13 AM
>
> with sendmsg playback send: loops: -1
>
> 2008/12/6 Faisal Maqsoodi
>  <faisalmaqsoodi at yahoo.com>:
>> Hi,
>>     Is there any built-in function, like playback, which plays a file
> again
>> and again unless interrupted. I want to use a simple function not FIFO.
>>
>> Faisal
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Sat Dec  6 08:37:22 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 6 Dec 2008 08:37:22 -0800
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <5e414ed0812060813k4877cc1anbae6a6fc2df9e4ba@mail.gmail.com>
References: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>
	<297073.11382.qm@web30701.mail.mud.yahoo.com>
	<5e414ed0812060813k4877cc1anbae6a6fc2df9e4ba@mail.gmail.com>
Message-ID: <09F35DA5-6613-4E98-A7D2-960BC24468C3@freeswitch.org>

Faisal,
Dennis makes a good point: you are mixing event socket syntax with  
dialplan syntax. I recommend starting with the dialplan example on the  
wiki. To get a single sound file to play over and over put it in a  
directory by itself.

Also, there is an undocumented feature called ".loc files" that I am  
researching. I believe this feature might give you more options. As  
soon as I can test it and document it I will report back. If any  
community members are using .loc files please let me know how they are  
working for you.

Thanks,
MC

Sent from my iPhone

On Dec 6, 2008, at 8:13 AM, Dennis <odermann at googlemail.com> wrote:

> although i do everything with socket outbound and php and not with the
> xml-dialplans, i feel, that you are missing the basics.
>
> first you should find out how to make a basic dialplan and how fs is
> working. then you can start with playing soundfiles and then, how to
> do a looped playback and other nice built in features.
> there are lots of dialplan samples delivered with fs and the wiki will
> help you to start with the rest.
>
> dennis
>
> 2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
>> I need some more help. I used send msg this way. Is there anything  
>> missing
>> bcoz its not working. Plz let me know what else should i do. When i  
>> dial
>> 2003, file is not played.
>> <extension name="Music">
>> sendmsg 2003
>> call-command: execute
>> execute-app-name: playback
>> execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav
>> loops: -1
>>  </extension>
>>
>> --- On Sat, 12/6/08, Dennis <odermann at googlemail.com> wrote:
>>
>> From: Dennis <odermann at googlemail.com>
>> Subject: Re: [Freeswitch-users] Playing a file again and again
>> To: freeswitch-users at lists.freeswitch.orgi
>> Date: Saturday, December 6, 2008, 2:13 AM
>>
>> with sendmsg playback send: loops: -1
>>
>> 2008/12/6 Faisal Maqsoodi
>> <faisalmaqsoodi at yahoo.com>:
>>> Hi,
>>>    Is there any built-in function, like playback, which plays a file
>> again
>>> and again unless interrupted. I want to use a simple function not  
>>> FIFO.
>>>
>>> Faisal
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Sat Dec  6 08:54:18 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Dec 2008 10:54:18 -0600
Subject: [Freeswitch-users] Playing a file again and again
In-Reply-To: <09F35DA5-6613-4E98-A7D2-960BC24468C3@freeswitch.org>
References: <5e414ed0812060213t5ce3f1a4ndcf21edb9b294217@mail.gmail.com>
	<297073.11382.qm@web30701.mail.mud.yahoo.com>
	<5e414ed0812060813k4877cc1anbae6a6fc2df9e4ba@mail.gmail.com>
	<09F35DA5-6613-4E98-A7D2-960BC24468C3@freeswitch.org>
Message-ID: <191c3a030812060854k17f59747y6e6e52e4b14175d8@mail.gmail.com>

you can set up an instance of mod_local_stream to endlessly play back a file
and many channels
can listen to it at the same time.

see ext 9999 in the default config.


On Sat, Dec 6, 2008 at 10:37 AM, Michael S Collins <msc at freeswitch.org>wrote:

> Faisal,
> Dennis makes a good point: you are mixing event socket syntax with
> dialplan syntax. I recommend starting with the dialplan example on the
> wiki. To get a single sound file to play over and over put it in a
> directory by itself.
>
> Also, there is an undocumented feature called ".loc files" that I am
> researching. I believe this feature might give you more options. As
> soon as I can test it and document it I will report back. If any
> community members are using .loc files please let me know how they are
> working for you.
>
> Thanks,
> MC
>
> Sent from my iPhone
>
> On Dec 6, 2008, at 8:13 AM, Dennis <odermann at googlemail.com> wrote:
>
> > although i do everything with socket outbound and php and not with the
> > xml-dialplans, i feel, that you are missing the basics.
> >
> > first you should find out how to make a basic dialplan and how fs is
> > working. then you can start with playing soundfiles and then, how to
> > do a looped playback and other nice built in features.
> > there are lots of dialplan samples delivered with fs and the wiki will
> > help you to start with the rest.
> >
> > dennis
> >
> > 2008/12/6 Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>:
> >> I need some more help. I used send msg this way. Is there anything
> >> missing
> >> bcoz its not working. Plz let me know what else should i do. When i
> >> dial
> >> 2003, file is not played.
> >> <extension name="Music">
> >> sendmsg 2003
> >> call-command: execute
> >> execute-app-name: playback
> >> execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav
> >> loops: -1
> >>  </extension>
> >>
> >> --- On Sat, 12/6/08, Dennis <odermann at googlemail.com> wrote:
> >>
> >> From: Dennis <odermann at googlemail.com>
> >> Subject: Re: [Freeswitch-users] Playing a file again and again
> >> To: freeswitch-users at lists.freeswitch.orgi
> >> Date: Saturday, December 6, 2008, 2:13 AM
> >>
> >> with sendmsg playback send: loops: -1
> >>
> >> 2008/12/6 Faisal Maqsoodi
> >> <faisalmaqsoodi at yahoo.com>:
> >>> Hi,
> >>>    Is there any built-in function, like playback, which plays a file
> >> again
> >>> and again unless interrupted. I want to use a simple function not
> >>> FIFO.
> >>>
> >>> Faisal
> >>>
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From frank at impactfax.com  Sat Dec  6 10:19:33 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 6 Dec 2008 13:19:33 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>
Message-ID: <04ab01c957cf$308e78e0$33014c0a@ws4>

Same thing with version 10640 build.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Saturday, December 06, 2008 1:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call

make current or install current svn on a different box.

/b

On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:

>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From uv at yuvalhertzog.com  Sat Dec  6 22:15:19 2008
From: uv at yuvalhertzog.com (UV)
Date: Sun, 7 Dec 2008 17:15:19 +1100
Subject: [Freeswitch-users] VoIP Product/Service and Man/Woman of the year
	2008
Message-ID: <F20429BF4E384EE692D8EBA33A398C9C@UVix>

Biz-news.com
<http://www.linkedin.com/redirect?url=http%3A%2F%2FBiz-news%2Ecom&urlhash=oN
Bl&_t=disc_detail_link>  is conducting a survey to discover the leading VoIP
man/woman of the year 2008, and the best VoIP product or service also for
the year 2008 . The winner will be selected based on the opinions of
professionals and technology enthusiasts in the industry. 

A) best VoIP product or service of 2008 
form: http://voip.biz-news.com/forms/py2008
<http://www.linkedin.com/redirect?url=http%3A%2F%2Fvoip%2Ebiz-news%2Ecom%2Ff
orms%2Fpy2008&urlhash=dspi&_t=disc_detail_link>  

B) VoIP Man/Woman of the year 2008 
form at: http://voip.biz-news.com/forms/my2008
<http://www.linkedin.com/redirect?url=http%3A%2F%2Fvoip%2Ebiz-news%2Ecom%2Ff
orms%2Fmy2008&urlhash=eDrg&_t=disc_detail_link>  

The results will be published early 2009 and share the raw data with the
community.

 

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From astmac at stillnewt.org  Sat Dec  6 22:49:44 2008
From: astmac at stillnewt.org (Martin Joseph)
Date: Sat, 6 Dec 2008 22:49:44 -0800
Subject: [Freeswitch-users] rootkit?
Message-ID: <723A7F88-B907-4C87-97E5-656D4F520272@stillnewt.org>

What is the rootkit item that appears to be added to the SVN of trunk?

Thanks,
Marty




From krice at suspicious.org  Sat Dec  6 22:59:14 2008
From: krice at suspicious.org (Ken Rice)
Date: Sun, 07 Dec 2008 00:59:14 -0600
Subject: [Freeswitch-users] rootkit?
In-Reply-To: <723A7F88-B907-4C87-97E5-656D4F520272@stillnewt.org>
Message-ID: <C560D062.4AE6A%krice@suspicious.org>

Its not really a rootkit... Its installs some screen and emacs profiles that
the FreeSwitch Dev Team use all the time...


> From: Martin Joseph <astmac at stillnewt.org>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 6 Dec 2008 22:49:44 -0800
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] rootkit?
> 
> What is the rootkit item that appears to be added to the SVN of trunk?
> 
> Thanks,
> Marty
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From jan.kubr at gmail.com  Sun Dec  7 03:31:26 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Sun, 7 Dec 2008 12:31:26 +0100
Subject: [Freeswitch-users] Read app ignores custom variable when called via
	socket interface
Message-ID: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>

Hi,
I checked out the current trunk (rev 10641) and found out that the
read app ignores the varname parameter, it always puts the result in
the DTMF-Digit variable. I'm calling it via the socket interface:

sendmsg
call-command: execute
execute-app-name: read
execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 #
event-lock:true


In the XML dialplan it works fine:

<action application="read" data="1 1
sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
<action application="phrase" data="spell,${res}"/>


I have been using the socket call above successfully in the 1.0.1 release.

Any ideas? Thanks,

Jan Kubr



From anthony.minessale at gmail.com  Sun Dec  7 11:44:03 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 7 Dec 2008 13:44:03 -0600
Subject: [Freeswitch-users] Read app ignores custom variable when called
	via socket interface
In-Reply-To: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
References: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
Message-ID: <191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>

That seems unlikely.
You sure about that?

The var param is in the middle of the data which is passed as 1 giant string
to the same exact app execution code.
I don't see how it could differentiate

did you try executing the info app right after to see all the vars.

I'm not saying i don't believe you but it seems fishy.



On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr <jan.kubr at gmail.com> wrote:

> Hi,
> I checked out the current trunk (rev 10641) and found out that the
> read app ignores the varname parameter, it always puts the result in
> the DTMF-Digit variable. I'm calling it via the socket interface:
>
> sendmsg
> call-command: execute
> execute-app-name: read
> execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 #
> event-lock:true
>
>
> In the XML dialplan it works fine:
>
> <action application="read" data="1 1
> sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
> <action application="phrase" data="spell,${res}"/>
>
>
> I have been using the socket call above successfully in the 1.0.1 release.
>
> Any ideas? Thanks,
>
> Jan Kubr
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Sun Dec  7 12:28:28 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 7 Dec 2008 14:28:28 -0600
Subject: [Freeswitch-users] Handling directory of sound files
In-Reply-To: <427787.87769.qm@web30703.mail.mud.yahoo.com>
References: <B25E3A72-98AE-4013-AAE0-CE978FD28717@freeswitch.org>
	<427787.87769.qm@web30703.mail.mud.yahoo.com>
Message-ID: <191c3a030812071228recd261eod1146fa3e731e613@mail.gmail.com>

mod_localstream is meant to be a moh source.
Playing a file endlessly can be also just done with extension logic

Make an extension that calls playback then transfers the call back to the
same extension.


On Sat, Dec 6, 2008 at 1:42 AM, Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>wrote:

> Thank u so much. mod local stream really works to play sound files from a
> local directory sequentially. Now can i jump to a specific file skipping the
> others? What should i use in dialplan? Have u any idea. Nothing is mentioned
> in the doc of mod lacal stream about that.
>                                                        Faisal
>
> --- On *Fri, 12/5/08, Michael S Collins <msc at freeswitch.org>* wrote:
>
> From: Michael S Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] Handling directory of sound files
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Date: Friday, December 5, 2008, 7:08 AM
>
> Check out mod_localstream on the wiki and see if that sounds like what you
> need. I'm still learning it all myself but I believe that's where you should
> start. Please report back with any questions and we will take it from there!
>
> -MC
>
>
> On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi < <faisalmaqsoodi at yahoo.com>
> faisalmaqsoodi at yahoo.com> wrote:
>
> Its not without music on hold completely. Say, e.g, moh is being played but
> when i press 1 it should start playing files contained in a specific
> directory sequentially or randomly. I havent got any solution to this
> problem yet. Can anyone plz guide me to some documentation or anything else
> regarding this matter.
>
>                                      Faisal
>
> _______________________________________________
> Freeswitch-users mailing list
> <Freeswitch-users at lists.freeswitch.org>
> Freeswitch-users at lists.freeswitch.org
> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users><http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> <http://www.freeswitch.org>http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Mon Dec  8 01:28:08 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 10:28:08 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
Message-ID: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>

hi,

we are fighting with two flaws in fs and would be happy, if they could
be fixed. we are using socket outbound.

1.) hangup a call in ringing state:
this worked in one of the last fs versions, but suddenly does not work anymore.

let's say, we have an inbound call and do 3 originates to different
targets. all 3 targets are in ringing state. the target, which answers
first, will be bridged with the inbound call, the other two (still
ringing) targets should be hung up.

we do not want fs to hang up the other two originates automatically.
we want to hang up the other two originates by sending the hangups. we
set "hangup_after_bridge=false" and "park_after_bridge=true".
we do sendmsg hangup uuid. but the originates are first hung up, when
they are answered. when they are in ringing state, the hangup will do
nothing (anymore).

as i said, it worked before, so i assume, that something has changed
in the latest trunks.



2.) sendmsg uuid *whatever* can cause to excute the command on the wrong uuid:

let's say, we have an inbound call and an outbound call - at least we
thing we have it ;-).
now we do for example sendmsg outbound_uuid hangup to hangup the
outbound call. but, if the uuid of the outbound call does not exist
(because there was a problem or something), the inbound will be hung
up instead.
the same happens with all sent messages to an uuid, which does not
exist. if we want to do a playback for the same outbound, the inbound
will hear it, if the outbound_uuid does not exist.

perhaps this is a feature, but i think that it would be nicer and more
reliable, if the sendmsg is only executed on the given uuid. if the
given uuid does not exist, nothing should happen or even nicer, an
event with an error should be sent to the socket.


thanks
dennis



From odermann at googlemail.com  Mon Dec  8 01:47:46 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 10:47:46 +0100
Subject: [Freeswitch-users] Two nice to have features in fs
Message-ID: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>

hi,

we are using socket outbound and found out, that there are two
(perhaps) small things, we would like to see in fs.

1.) if we try to playback a soundfile, which does not exist, we do not
get an error or something. in the cli of fs we can see the error, but
we do not get anything about it over the socket.

we get an execute event for the playback and directly after it an
execute-complete.
wouldn't it be nicer, if one could get a notice about an error, like:
execute-complete-error or a variable, which is set with an error, if
an error occours?


2.) the session heartbeat event is an absolutely great feature in fs.
but sometimes we would like to get the session heartbeat every 5
seconds. at the moment the allowed minimum is every 10 seconds. a
smaller setting, like every 5 seconds, will result in a session
heartbeat of every 60 seconds.

to help us out, we edit the switch_core_session.c at line 899 and
change the "seconds < 10" to "seconds < 5".

because one has to set "enable_heartbeat_events=5" manually, i do not
think, that there is a risk, that others, who do not want the
heartbeat to come that often, will be negatively affected by this
change.

might it be possible, to do the same changes to the default code?


thanks
dennis



From regs at kinetix.gr  Mon Dec  8 02:11:32 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Mon, 08 Dec 2008 12:11:32 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
Message-ID: <493CF2D4.6010904@kinetix.gr>

Any updates about the "which side hanged up" potential variable?

Michael S Collins wrote:
> Makes sense. I will look into this. 
> -MC
>
>
> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr 
> <mailto:regs at kinetix.gr>> wrote:
>
>> I am sending this second email to request/suggest/enquire about 
>> something relevant :
>>
>> Wouldn't it be useful to know which end of a specific call leg send 
>> the protocol
>> specific hangup cause? Otherwise it would be difficult to understand 
>> what really happened.
>>
>>
>>
>> Michael S Collins wrote:
>>> I will do some research on this and let you know what I find out.  
>>> Question: are these internal calls or pstn or ?? Just curious about  
>>> your environment.
>>>
>>> Thanks,
>>> MC
>>>
>>>
>>>
>>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr <mailto:regs at kinetix.gr>>  
>>> wrote:
>>>
>>>   
>>>> The proto_specific_hangup_cause is missing on one of the two
>>>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>> When the callee hangs up it is missing from the b-leg CDR. Is this  
>>>> nornal?
>>>>
>>>> And another question : what piece of info could inform me about who
>>>> hanged up?
>>>>
>>>>
>>>> -- 
>>>> -------------------------------------------
>>>> Apostolos Pantsiopoulos
>>>> Kinetix Tele.com R & D
>>>> email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>> -------------------------------------------
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>     
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>   
>>
>>
>> -- 
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>> ------------------------------------------- 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org 
>> <mailto:Freeswitch-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From jonas.gauffin at gmail.com  Mon Dec  8 02:36:26 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 8 Dec 2008 11:36:26 +0100
Subject: [Freeswitch-users] Autoanswer
In-Reply-To: <d021d3be0811260827o68efc7drf2985ad684c72b71@mail.gmail.com>
References: <d021d3be0811260720p29736159p953b6bdd5bef6621@mail.gmail.com>
	<191c3a030811260819l5081b3a2q9b606d7a109d58be@mail.gmail.com>
	<d021d3be0811260827o68efc7drf2985ad684c72b71@mail.gmail.com>
Message-ID: <d021d3be0812080236v4c7bb4b4he8ab7faae02c09f0@mail.gmail.com>

I've tried what you said, and both legs still get auto answered.

My string:
api originate
{gate_user_id=157,gate_site_id=87,origination_caller_id_name='Namie
Amuro'}[sip_invite_params=intercom=true,sip_h_Call-Info=<sip:$${domain}>;answer-after=0,sip_auto_answer=true]sofia/localdomain/u1000157%
192.168.1.111 '&execute_extension(1202 XML)'


On Wed, Nov 26, 2008 at 5:27 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Ahh. gr8. thanks.
>
>
> On Wed, Nov 26, 2008 at 5:19 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you can put vars in [ ] before each channel in the list to apply them only
>> to that one channel.
>>
>> {global=true}[only_this_channel=true]sofia/foo/foo at bar.com
>> ,[only_this_channel_again=true]sofia/foo/baz at bar.com
>>
>>
>> On Wed, Nov 26, 2008 at 9:20 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello
>>> I send an API command through the event socket that looks like this (the
>>> first two variables are used by our server):
>>>
>>> api originate
>>> {gate_user_id=44,gate_site_id=1,sip_invite_params=intercom=true,sip_h_Call-Info=<sip:$${domain}>;answer-after=0,sip_auto_answer=true,origination_caller_id_name='Jonas
>>> Gauffin',origination_caller_id_number=+4623666XXXX,sip_auto_answer=true}sofia/localdomain/u1000044%
>>> 192.168.1.112 '&execute_extension(8901 XML)'
>>>
>>> The command works just as it should. The problem is that the auto-answer
>>> variables seems to stick to the b-leg (execute extensions), which means that
>>> both calls gets auto answered. What I want is that only the first call gets
>>> answered.
>>>
>>> (A logged in user presses the "call" icon in our webdirectory, which
>>> makes freeswitch use his phone to call 8901)
>>>
>>> //Jonas
>>>
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jan.kubr at gmail.com  Mon Dec  8 02:48:30 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Mon, 8 Dec 2008 11:48:30 +0100
Subject: [Freeswitch-users] Read app ignores custom variable when called
	via socket interface
In-Reply-To: <191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>
References: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
	<191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>
Message-ID: <698401620812080248u5f866d7es950d0019686b2c99@mail.gmail.com>

OK my bad. The variable is set (I can see it in the Freeswitch console
when I use the info app), but they are only not send to me via the
socket interface. I get the "variable_*" variables only in the
beginning (after calling connect), but not in the events. How do I
enable this?
Thanks,
Jan

On Sun, Dec 7, 2008 at 8:44 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> That seems unlikely.
> You sure about that?
>
> The var param is in the middle of the data which is passed as 1 giant string
> to the same exact app execution code.
> I don't see how it could differentiate
>
> did you try executing the info app right after to see all the vars.
>
> I'm not saying i don't believe you but it seems fishy.
>
>
>
> On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr <jan.kubr at gmail.com> wrote:
>>
>> Hi,
>> I checked out the current trunk (rev 10641) and found out that the
>> read app ignores the varname parameter, it always puts the result in
>> the DTMF-Digit variable. I'm calling it via the socket interface:
>>
>> sendmsg
>> call-command: execute
>> execute-app-name: read
>> execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 #
>> event-lock:true
>>
>>
>> In the XML dialplan it works fine:
>>
>> <action application="read" data="1 1
>> sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
>> <action application="phrase" data="spell,${res}"/>
>>
>>
>> I have been using the socket call above successfully in the 1.0.1 release.
>>
>> Any ideas? Thanks,
>>
>> Jan Kubr
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From carole.olivier at enst.fr  Mon Dec  8 05:10:16 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Mon, 8 Dec 2008 05:10:16 -0800 (PST)
Subject: [Freeswitch-users] conference configured to call automatically
 the attended does not work
In-Reply-To: <20856465.post@talk.nabble.com>
References: <20856465.post@talk.nabble.com>
Message-ID: <20895038.post@talk.nabble.com>


I have found my mistake.

In the dialplan I have written <action
application="conference_set_auto_outcall" data="sofia/internal/.../> instead
of <action application="conference_set_auto_outcall" data="user/.../> 

Sorry for the post,
Carole


Carole O. wrote:
> 
> Hello,
> 
> I have got some problems for the configuration of a simple conference
> which should be established by calling an extension and automatically
> inviting 2 people.
> 
> Actually, this is based on the default configuration of Freeswitch
> (extension 0911). I have changed it a little:
> 
> <extension name="conference_test">
>     <condition field="destination_number" expression="^0911$">
>     <!--  These parameters effect the outcalls made once you join 
>       --> 
>       <action application="set"
> data="conference_auto_outcall_caller_id_name=telephoneX" /> 
>       <action application="set"
> data="conference_auto_outcall_id_number=0911" /> 
>       <action application="set" data="conference_auto_outcall_timeout=60"
> /> 
>       <action application="set" data="conference_auto_outcall_flags=none"
> /> 
>  
>     <!--  list of devices called  --> 
>       <action application="conference_set_auto_outcall"
> data="sofia/internal/1010@$${domain}" /> 
>       <action application="conference_set_auto_outcall"
> data="sofia/internal/1002@$${domain}" /> 
> 
>       <action application="conference" data="conferenceTest at default" /> 
>     </condition>
>  </extension>
> 
> 
> I have attached a file with the console errors. There are some errors (moh
> errors) but since these were also present for room conference and it did
> not prevent it for working, I guess this is not the fundamental reason for
> the previous problem.
> 
> I have an additional question. I have installed freeswitch from
> opensuse.org, there is a simple "one-click installation" but I am not sure
> this was a good idea, it seems to be light isn't?
> 
> Thanks for your help,
> Carole
>  http://www.nabble.com/file/p20856465/error_console.txt error_console.txt 
> 

-- 
View this message in context: http://www.nabble.com/conference-configured-to-call-automatically-the-attended-does-not-work-tp20856465p20895038.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From mike at jerris.com  Mon Dec  8 05:33:01 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 8 Dec 2008 08:33:01 -0500
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
Message-ID: <2BA1F89F-C5A8-4DC6-B297-052363CC178B@jerris.com>

Can you please file bugs on http://jira.freeswitch.org with full sip  
trace and FreeSWITCH debug output of these cases.


On Dec 8, 2008, at 4:28 AM, Dennis wrote:

> hi,
>
> we are fighting with two flaws in fs and would be happy, if they could
> be fixed. we are using socket outbound.
>
> 1.) hangup a call in ringing state:
> this worked in one of the last fs versions, but suddenly does not  
> work anymore.
>
> let's say, we have an inbound call and do 3 originates to different
> targets. all 3 targets are in ringing state. the target, which answers
> first, will be bridged with the inbound call, the other two (still
> ringing) targets should be hung up.
>
> we do not want fs to hang up the other two originates automatically.
> we want to hang up the other two originates by sending the hangups. we
> set "hangup_after_bridge=false" and "park_after_bridge=true".
> we do sendmsg hangup uuid. but the originates are first hung up, when
> they are answered. when they are in ringing state, the hangup will do
> nothing (anymore).
>
> as i said, it worked before, so i assume, that something has changed
> in the latest trunks.
>
>
>
> 2.) sendmsg uuid *whatever* can cause to excute the command on the  
> wrong uuid:
>
> let's say, we have an inbound call and an outbound call - at least we
> thing we have it ;-).
> now we do for example sendmsg outbound_uuid hangup to hangup the
> outbound call. but, if the uuid of the outbound call does not exist
> (because there was a problem or something), the inbound will be hung
> up instead.
> the same happens with all sent messages to an uuid, which does not
> exist. if we want to do a playback for the same outbound, the inbound
> will hear it, if the outbound_uuid does not exist.
>
> perhaps this is a feature, but i think that it would be nicer and more
> reliable, if the sendmsg is only executed on the given uuid. if the
> given uuid does not exist, nothing should happen or even nicer, an
> event with an error should be sent to the socket.
>
>
> thanks
> dennis
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Mon Dec  8 05:34:42 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 8 Dec 2008 08:34:42 -0500
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
Message-ID: <1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>


On Dec 8, 2008, at 4:47 AM, Dennis wrote:

> at the moment the allowed minimum is every 10 seconds. a
> smaller setting, like every 5 seconds, will result in a session
> heartbeat of every 60 seconds.

Huh?



From anthony.minessale at gmail.com  Mon Dec  8 05:48:02 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 07:48:02 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493CF2D4.6010904@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
	<493CF2D4.6010904@kinetix.gr>
Message-ID: <191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>

sip_hangup_disposition will be set to recv_bye on the side that was hungup.


On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  Any updates about the "which side hanged up" potential variable?
>
> Michael S Collins wrote:
>
> Makes sense. I will look into this.
> -MC
>
>
> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
> wrote:
>
>  I am sending this second email to request/suggest/enquire about something
> relevant :
>
> Wouldn't it be useful to know which end of a specific call leg send the
> protocol
> specific hangup cause? Otherwise it would be difficult to understand what
> really happened.
>
>
>
> Michael S Collins wrote:
>
> I will do some research on this and let you know what I find out.
> Question: are these internal calls or pstn or ?? Just curious about
> your environment.
>
> Thanks,
> MC
>
>
>
> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <regs at kinetix.gr>regs at kinetix.gr>
> wrote:
>
>
>
>  The proto_specific_hangup_cause is missing on one of the two
> call legs. When the caller hangs up it is missing from the a-leg CDR.
> When the callee hangs up it is missing from the b-leg CDR. Is this
> nornal?
>
> And another question : what piece of info could inform me about who
> hanged up?
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>  _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>   _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Dec  8 06:04:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 08:04:51 -0600
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
Message-ID: <191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>

#2 was because when you sendmsg with no uuid on an outbound socket it
defaults to the session who called you.
I changed to code to make a distinction between not supplying a uuid and
supplying an invalid uuid.

#1 seems hard to believe.  Please provide a console trace of the channel
*ignoring* the hangup command.


On Mon, Dec 8, 2008 at 3:28 AM, Dennis <odermann at googlemail.com> wrote:

> hi,
>
> we are fighting with two flaws in fs and would be happy, if they could
> be fixed. we are using socket outbound.
>
> 1.) hangup a call in ringing state:
> this worked in one of the last fs versions, but suddenly does not work
> anymore.
>
> let's say, we have an inbound call and do 3 originates to different
> targets. all 3 targets are in ringing state. the target, which answers
> first, will be bridged with the inbound call, the other two (still
> ringing) targets should be hung up.
>
> we do not want fs to hang up the other two originates automatically.
> we want to hang up the other two originates by sending the hangups. we
> set "hangup_after_bridge=false" and "park_after_bridge=true".
> we do sendmsg hangup uuid. but the originates are first hung up, when
> they are answered. when they are in ringing state, the hangup will do
> nothing (anymore).
>
> as i said, it worked before, so i assume, that something has changed
> in the latest trunks.
>
>
>
> 2.) sendmsg uuid *whatever* can cause to excute the command on the wrong
> uuid:
>
> let's say, we have an inbound call and an outbound call - at least we
> thing we have it ;-).
> now we do for example sendmsg outbound_uuid hangup to hangup the
> outbound call. but, if the uuid of the outbound call does not exist
> (because there was a problem or something), the inbound will be hung
> up instead.
> the same happens with all sent messages to an uuid, which does not
> exist. if we want to do a playback for the same outbound, the inbound
> will hear it, if the outbound_uuid does not exist.
>
> perhaps this is a feature, but i think that it would be nicer and more
> reliable, if the sendmsg is only executed on the given uuid. if the
> given uuid does not exist, nothing should happen or even nicer, an
> event with an error should be sent to the socket.
>
>
> thanks
> dennis
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From regs at kinetix.gr  Mon Dec  8 06:13:47 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Mon, 08 Dec 2008 16:13:47 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
Message-ID: <493D2B9B.6050205@kinetix.gr>

Not necessarily. For instance I got a "send_cancel" when the
calling party hanged up before the other party could pick up.
Also, shouldn't something like that be protocol/technology
neutral?



Anthony Minessale wrote:
> sip_hangup_disposition will be set to recv_bye on the side that was 
> hungup.
>
>
> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     Any updates about the "which side hanged up" potential variable?
>
>     Michael S Collins wrote:
>>     Makes sense. I will look into this. 
>>     -MC
>>
>>
>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>>
>>>     I am sending this second email to request/suggest/enquire about
>>>     something relevant :
>>>
>>>     Wouldn't it be useful to know which end of a specific call leg
>>>     send the protocol
>>>     specific hangup cause? Otherwise it would be difficult to
>>>     understand what really happened.
>>>
>>>
>>>
>>>     Michael S Collins wrote:
>>>>     I will do some research on this and let you know what I find out.  
>>>>     Question: are these internal calls or pstn or ?? Just curious about  
>>>>     your environment.
>>>>
>>>>     Thanks,
>>>>     MC
>>>>
>>>>
>>>>
>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <mailto:regs at kinetix.gr>regs at kinetix.gr <mailto:regs at kinetix.gr>>  
>>>>     wrote:
>>>>
>>>>       
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this  
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     -- 
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>         
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>       
>>>
>>>
>>>     -- 
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     ------------------------------------------- 
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org
>>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From steveayre at gmail.com  Mon Dec  8 05:42:14 2008
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 8 Dec 2008 13:42:14 +0000
Subject: [Freeswitch-users] rootkit?
In-Reply-To: <723A7F88-B907-4C87-97E5-656D4F520272@stillnewt.org>
References: <723A7F88-B907-4C87-97E5-656D4F520272@stillnewt.org>
Message-ID: <eafa772f0812080542p417cf9ddu12867963d6892aeb@mail.gmail.com>

A few files which you can choose to install to let the Freeswitch
developers access your machine remotely if you ask them to look at a
problem you're having with Freeswitch.

It's not a real rootkit - just the SSH public key for their private
key (so they don't need a password) and configuration files for bash +
emacs (so they're working in the environment they're used to). It
doesn't burrow into your OS, hide itself or any of the other insidious
things a real rootkit does, and once you delete the files they can no
longer access your machine.

It isn't installed unless you choose to do so.

-Steve



2008/12/7 Martin Joseph <astmac at stillnewt.org>:
> What is the rootkit item that appears to be added to the SVN of trunk?
>
> Thanks,
> Marty
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From odermann at googlemail.com  Mon Dec  8 07:18:18 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 16:18:18 +0100
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
	<1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
Message-ID: <5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>

> Huh?

src/switch_core_session.c vom line 899 to 901:

	if (seconds < 10) {
		seconds = 60;
	}



From anthony.minessale at gmail.com  Mon Dec  8 07:30:15 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 09:30:15 -0600
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
	<1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
	<5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>
Message-ID: <191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>

done

On Mon, Dec 8, 2008 at 9:18 AM, Dennis <odermann at googlemail.com> wrote:

> > Huh?
>
> src/switch_core_session.c vom line 899 to 901:
>
>        if (seconds < 10) {
>                seconds = 60;
>         }
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Dec  8 07:42:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 09:42:51 -0600
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
	<1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
	<5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>
	<191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>
Message-ID: <191c3a030812080742w7e6e2a3co42453a7b41d88826@mail.gmail.com>

both done

On Mon, Dec 8, 2008 at 9:30 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> done
>
>
> On Mon, Dec 8, 2008 at 9:18 AM, Dennis <odermann at googlemail.com> wrote:
>
>> > Huh?
>>
>> src/switch_core_session.c vom line 899 to 901:
>>
>>        if (seconds < 10) {
>>                seconds = 60;
>>         }
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From odermann at googlemail.com  Mon Dec  8 07:47:38 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 16:47:38 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
Message-ID: <5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>

> #2 was because when you sendmsg with no uuid on an outbound socket it
> defaults to the session who called you.
> I changed to code to make a distinction between not supplying a uuid and
> supplying an invalid uuid.

anthony, thanks for the quick reaction!

we just tested you changes and it works the opposite way it should.

this means: when we do not send an uuid, we get an an error
(Reply-Text => -ERR invalid session id []). if we send a wrong/not
existing uuid, the command will be executed on the inbound uuid.




> #1 seems hard to believe.  Please provide a console trace of the channel
> *ignoring* the hangup command.

i know it is hard to believe, we didn't believe it either ;-)

have a look at http://pastebin.freeswitch.org/6367

what we simply do here: the inbound is coming in, then we do an
originate and hang up the inbound. then we directly send a hangup for
the outbound. the outbound will go on ringing.
then, when the ringing outbound is answered, we directly get the hangup.
fs gets the hangup and remembers it, but seems to wait till the answer
to execute this command.



From anthony.minessale at gmail.com  Mon Dec  8 07:52:17 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 09:52:17 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493D2B9B.6050205@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
	<493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
	<493D2B9B.6050205@kinetix.gr>
Message-ID: <191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>

it is protocol neutral, that's why it starts with sip_

the variable can be any of:

send_bye
recv_bye
send_cancel
send_refuse


using that value you can determine the information you asked.  I answered
your specific question which was:
determining "which side hanged up".  Maybe you should beat around the bush
less with your "requirements" for your application you are expecting me to
support for you.

I already added 2 patches for you right.  Just be clear about what you want.



On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>  Not necessarily. For instance I got a "send_cancel" when the
> calling party hanged up before the other party could pick up.
> Also, shouldn't something like that be protocol/technology
> neutral?
>
>
>
> Anthony Minessale wrote:
>
> sip_hangup_disposition will be set to recv_bye on the side that was hungup.
>
>
> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:
>
>>  Any updates about the "which side hanged up" potential variable?
>>
>> Michael S Collins wrote:
>>
>>  Makes sense. I will look into this.
>> -MC
>>
>>
>> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>> wrote:
>>
>>  I am sending this second email to request/suggest/enquire about
>> something relevant :
>>
>> Wouldn't it be useful to know which end of a specific call leg send the
>> protocol
>> specific hangup cause? Otherwise it would be difficult to understand what
>> really happened.
>>
>>
>>
>> Michael S Collins wrote:
>>
>> I will do some research on this and let you know what I find out.
>> Question: are these internal calls or pstn or ?? Just curious about
>> your environment.
>>
>> Thanks,
>> MC
>>
>>
>>
>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <regs at kinetix.gr>regs at kinetix.gr>
>> wrote:
>>
>>
>>
>>  The proto_specific_hangup_cause is missing on one of the two
>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>> When the callee hangs up it is missing from the b-leg CDR. Is this
>> nornal?
>>
>> And another question : what piece of info could inform me about who
>> hanged up?
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>  _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>   _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>  ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Mon Dec  8 08:07:01 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 17:07:01 +0100
Subject: [Freeswitch-users] Two nice to have features in fs
In-Reply-To: <191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>
References: <5e414ed0812080147o62eda1d5r459ce1036962780a@mail.gmail.com>
	<1E1A1007-46F4-4CE1-B092-DEFE4F2DC20A@jerris.com>
	<5e414ed0812080718w1c45c155l777f4c0342b4eb5@mail.gmail.com>
	<191c3a030812080730w694c5c50u295d0f1522cdb0d2@mail.gmail.com>
Message-ID: <5e414ed0812080807w609e56f4o3c380f5417e32010@mail.gmail.com>

great, that works! thanks a lot!

just tested the changes according an error, when a file is missing.
thanks again!


2008/12/8 Anthony Minessale <anthony.minessale at gmail.com>:
> done
>
> On Mon, Dec 8, 2008 at 9:18 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> > Huh?
>>
>> src/switch_core_session.c vom line 899 to 901:
>>
>>        if (seconds < 10) {
>>                seconds = 60;
>>        }
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Mon Dec  8 08:08:48 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 10:08:48 -0600
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
Message-ID: <191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>

try the sendmsg issue again

are you doing the hangup with

api uuid_kill <uuid>



On Mon, Dec 8, 2008 at 9:47 AM, Dennis <odermann at googlemail.com> wrote:

> > #2 was because when you sendmsg with no uuid on an outbound socket it
> > defaults to the session who called you.
> > I changed to code to make a distinction between not supplying a uuid and
> > supplying an invalid uuid.
>
> anthony, thanks for the quick reaction!
>
> we just tested you changes and it works the opposite way it should.
>
> this means: when we do not send an uuid, we get an an error
> (Reply-Text => -ERR invalid session id []). if we send a wrong/not
> existing uuid, the command will be executed on the inbound uuid.
>
>
>
>
> > #1 seems hard to believe.  Please provide a console trace of the channel
> > *ignoring* the hangup command.
>
> i know it is hard to believe, we didn't believe it either ;-)
>
> have a look at http://pastebin.freeswitch.org/6367
>
> what we simply do here: the inbound is coming in, then we do an
> originate and hang up the inbound. then we directly send a hangup for
> the outbound. the outbound will go on ringing.
> then, when the ringing outbound is answered, we directly get the hangup.
> fs gets the hangup and remembers it, but seems to wait till the answer
> to execute this command.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Mon Dec  8 08:19:02 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 17:19:02 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
Message-ID: <5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>

i have to shift places. will be back in a few minutes and test.

no, we are using the simple sendmsg uuid hangup. as far as we
remember, we do not use api uuid_kill, because we do not get a hangup
event with this.


2008/12/8 Anthony Minessale <anthony.minessale at gmail.com>:
> try the sendmsg issue again
>
> are you doing the hangup with
>
> api uuid_kill <uuid>



From anthony.minessale at gmail.com  Mon Dec  8 08:44:58 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 10:44:58 -0600
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
Message-ID: <191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>

you would get a hangup event in either case.


On Mon, Dec 8, 2008 at 10:19 AM, Dennis <odermann at googlemail.com> wrote:

> i have to shift places. will be back in a few minutes and test.
>
> no, we are using the simple sendmsg uuid hangup. as far as we
> remember, we do not use api uuid_kill, because we do not get a hangup
> event with this.
>
>
> 2008/12/8 Anthony Minessale <anthony.minessale at gmail.com>:
> > try the sendmsg issue again
> >
> > are you doing the hangup with
> >
> > api uuid_kill <uuid>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From odermann at googlemail.com  Mon Dec  8 08:56:15 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 17:56:15 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
Message-ID: <5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>

> you would get a hangup event in either case.

yes, you are right. we just tested and saw that. the reason for
sendmsg hangup, was the sometimes useful event-lock.

it works with api uuid_kill as we wanted. but with sendmsg hangup it
still does not work. shouldn't sendmsg hangup work like uuid_kill
here? how useful could it be, to let it ring, when the hangup was
already sent and is immediately executed when the anser is sent?


#2 now works perfectly. thanks for the great support!



From mrjoebain at gmail.com  Mon Dec  8 08:57:11 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Mon, 8 Dec 2008 16:57:11 +0000
Subject: [Freeswitch-users] Catching hangups
Message-ID: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>

Hi,

I'm writing an IVR in Lua and am having problems dealing with hangups
cleanly. Very often session:ready() reports true long after I have hung up
and the hangup hook function I have set doesn't get called either. It seems
to report that the session is active indefinitely in some cases where a loop
keeps trying to get some dtmf key presses. Is there any trick to using
session:ready() or the hangup hook that I might have missed?

On a slightly related point I can't seem to access the session properties,
e.g. session.caller_id_num has a value of nil. Any thoughts here?

Thanks in advance,

Joe Bain
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From anthony.minessale at gmail.com  Mon Dec  8 09:11:52 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 11:11:52 -0600
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
	<5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
Message-ID: <191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>

channels in originate were not checking for private events.
now they should but if send them commands to do crazy stuff like play a file
while they are
in the middle of originating there could be ill side effects (e.g. play file
before media was established etc which could cause the call to abort)

btw you can send

call-command: hangup
hangup-cause: normal_clearing

in place of
call-command: execute
execute-app-name: hangup
execute-app-arg: normal_clearing


On Mon, Dec 8, 2008 at 10:56 AM, Dennis <odermann at googlemail.com> wrote:

> > you would get a hangup event in either case.
>
> yes, you are right. we just tested and saw that. the reason for
> sendmsg hangup, was the sometimes useful event-lock.
>
> it works with api uuid_kill as we wanted. but with sendmsg hangup it
> still does not work. shouldn't sendmsg hangup work like uuid_kill
> here? how useful could it be, to let it ring, when the hangup was
> already sent and is immediately executed when the anser is sent?
>
>
> #2 now works perfectly. thanks for the great support!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Mon Dec  8 09:13:34 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Dec 2008 09:13:34 -0800
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
Message-ID: <87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>

Joe,

A few questions... what svn rev are you running? Which operating
system? Finally, is it possible for you to put your dialplan and Lua
script up at pastebin.freeswitch.org?

Thanks,
MC

On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
> Hi,
>
> I'm writing an IVR in Lua and am having problems dealing with hangups
> cleanly. Very often session:ready() reports true long after I have hung up
> and the hangup hook function I have set doesn't get called either. It seems
> to report that the session is active indefinitely in some cases where a loop
> keeps trying to get some dtmf key presses. Is there any trick to using
> session:ready() or the hangup hook that I might have missed?
>
> On a slightly related point I can't seem to access the session properties,
> e.g. session.caller_id_num has a value of nil. Any thoughts here?
>
> Thanks in advance,
>
> Joe Bain
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Mon Dec  8 09:18:01 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 11:18:01 -0600
Subject: [Freeswitch-users] Read app ignores custom variable when called
	via socket interface
In-Reply-To: <698401620812080248u5f866d7es950d0019686b2c99@mail.gmail.com>
References: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
	<191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>
	<698401620812080248u5f866d7es950d0019686b2c99@mail.gmail.com>
Message-ID: <191c3a030812080918x6acd7564q3247055cdc941641@mail.gmail.com>

i added a patch to index the variables on the
SWITCH_EVENT_CHANNEL_EXECUTE_COMPLETE
if you want to update

otherwise you can use uuid_getvar to retrieve the variable


On Mon, Dec 8, 2008 at 4:48 AM, Jan Kubr <jan.kubr at gmail.com> wrote:

> OK my bad. The variable is set (I can see it in the Freeswitch console
> when I use the info app), but they are only not send to me via the
> socket interface. I get the "variable_*" variables only in the
> beginning (after calling connect), but not in the events. How do I
> enable this?
> Thanks,
> Jan
>
> On Sun, Dec 7, 2008 at 8:44 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > That seems unlikely.
> > You sure about that?
> >
> > The var param is in the middle of the data which is passed as 1 giant
> string
> > to the same exact app execution code.
> > I don't see how it could differentiate
> >
> > did you try executing the info app right after to see all the vars.
> >
> > I'm not saying i don't believe you but it seems fishy.
> >
> >
> >
> > On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr <jan.kubr at gmail.com> wrote:
> >>
> >> Hi,
> >> I checked out the current trunk (rev 10641) and found out that the
> >> read app ignores the varname parameter, it always puts the result in
> >> the DTMF-Digit variable. I'm calling it via the socket interface:
> >>
> >> sendmsg
> >> call-command: execute
> >> execute-app-name: read
> >> execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000
> #
> >> event-lock:true
> >>
> >>
> >> In the XML dialplan it works fine:
> >>
> >> <action application="read" data="1 1
> >> sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
> >> <action application="phrase" data="spell,${res}"/>
> >>
> >>
> >> I have been using the socket call above successfully in the 1.0.1
> release.
> >>
> >> Any ideas? Thanks,
> >>
> >> Jan Kubr
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Mon Dec  8 09:26:52 2008
From: odermann at googlemail.com (Dennis)
Date: Mon, 8 Dec 2008 18:26:52 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
	<5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
	<191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
Message-ID: <5e414ed0812080926ob7a134emc4a1d00d447f87b8@mail.gmail.com>

thanks, now it works as we expected.

and thanks for the hint, how we should send the hangup with sendmsg.
we will do it your way :-)


2008/12/8 Anthony Minessale <anthony.minessale at gmail.com>:
> channels in originate were not checking for private events.
> now they should but if send them commands to do crazy stuff like play a file
> while they are
> in the middle of originating there could be ill side effects (e.g. play file
> before media was established etc which could cause the call to abort)
>
> btw you can send
>
> call-command: hangup
> hangup-cause: normal_clearing
>
> in place of
> call-command: execute
> execute-app-name: hangup
> execute-app-arg: normal_clearing
>
>
> On Mon, Dec 8, 2008 at 10:56 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> > you would get a hangup event in either case.
>>
>> yes, you are right. we just tested and saw that. the reason for
>> sendmsg hangup, was the sometimes useful event-lock.
>>
>> it works with api uuid_kill as we wanted. but with sendmsg hangup it
>> still does not work. shouldn't sendmsg hangup work like uuid_kill
>> here? how useful could it be, to let it ring, when the hangup was
>> already sent and is immediately executed when the anser is sent?
>>
>>
>> #2 now works perfectly. thanks for the great support!
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From dule.maillist at gmail.com  Mon Dec  8 09:51:29 2008
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 8 Dec 2008 12:51:29 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <04ab01c957cf$308e78e0$33014c0a@ws4>
References: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>
	<04ab01c957cf$308e78e0$33014c0a@ws4>
Message-ID: <914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>

>From my understanding, I didn't think tone_detect detects DTMF since it's
dual frequencies, rather tone_detect detects single frequencies like fax
tones.

I would just run an IVR with a session.read or session.getDigits to collect
DTMF.

Dan

On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com> wrote:

> Same thing with version 10640 build.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Saturday, December 06, 2008 1:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
>
> make current or install current svn on a different box.
>
> /b
>
> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>
> >
> > Ideas?  Am I doing something stupid or is tone_detect not just right
> > here?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Dec  8 09:54:40 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2008 11:54:40 -0600
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>
References: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>
	<04ab01c957cf$308e78e0$33014c0a@ws4>
	<914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>
Message-ID: <191c3a030812080954x6bbf0c85o209d157e6a05412e@mail.gmail.com>

tone_detect can detect an MF tone up to 6 at once. (in practice)


On Mon, Dec 8, 2008 at 11:51 AM, Dan Le <dule.maillist at gmail.com> wrote:

> From my understanding, I didn't think tone_detect detects DTMF since it's
> dual frequencies, rather tone_detect detects single frequencies like fax
> tones.
>
> I would just run an IVR with a session.read or session.getDigits to collect
> DTMF.
>
> Dan
>
> On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com>wrote:
>
>> Same thing with version 10640 build.
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Brian West
>> Sent: Saturday, December 06, 2008 1:01 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] key tone trigger event during call
>>
>> make current or install current svn on a different box.
>>
>> /b
>>
>> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>>
>> >
>> > Ideas?  Am I doing something stupid or is tone_detect not just right
>> > here?
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Mon Dec  8 09:56:30 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 8 Dec 2008 12:56:30 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>
References: <FB741C79-4B3A-4D8B-AB3A-435BEB5FA2EF@freeswitch.org>
	<04ab01c957cf$308e78e0$33014c0a@ws4>
	<914fc92a0812080951l11ea8910q7bcd9632c2c956b0@mail.gmail.com>
Message-ID: <F8B619FF-4F53-4374-B903-871DB1C938E4@jerris.com>

Are you really trying to detect a tone, or are you trying to detect  
dtmf (could be delivered via rfc2833, info, etc) ?

Mike

On Dec 8, 2008, at 12:51 PM, Dan Le wrote:

> From my understanding, I didn't think tone_detect detects DTMF since  
> it's dual frequencies, rather tone_detect detects single frequencies  
> like fax tones.
>
> I would just run an IVR with a session.read or session.getDigits to  
> collect DTMF.
> Dan
>
>
> On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com>  
> wrote:
> Same thing with version 10640 build.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Saturday, December 06, 2008 1:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
>
> make current or install current svn on a different box.
>
> /b
>
> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>
> >
> > Ideas?  Am I doing something stupid or is tone_detect not just right
> > here?
>

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From john at loopfx.com  Mon Dec  8 13:36:07 2008
From: john at loopfx.com (John Rutherford)
Date: Mon, 8 Dec 2008 16:36:07 -0500
Subject: [Freeswitch-users] No audio after transfer
Message-ID: <81469655CA61444CBB034826ABC6F6E331D5A3@anniesue.loop.local>

I'm trying to get an attended transfer work with freeSWITCH, but it's
not quite working.  I have Microsoft Speech Server on one side and
Televantage on the other.  

 

MSS is originating a call, which freeSWITCH is bridging to Televantage.
That calls connects just fine.  Then, MSS sends a re-INVITE to
Televantage to put the call on hold.  This works too.  Then, MSS
originates another call to freeSWITCH, which is again bridged to
Televantage.  This works fine too.  

 

Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
should be complete, but there is no audio between the two calls-just
silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not
seeing anything obviously wrong.  

 

After the REFER, I can see audio for both calls going between freeSWITCH
and Televantage, so it seems that the only thing missing is freeSWITCH
routing the audio from one call to the other call and vice-versa.

 

 

Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this.  

 

Thanks!

John

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From brian at freeswitch.org  Mon Dec  8 13:48:28 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Dec 2008 15:48:28 -0600
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E331D5A3@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D5A3@anniesue.loop.local>
Message-ID: <244F3A58-EA28-4A76-AD5E-160E65B5B421@freeswitch.org>

Are you on SVN trunk?  If not what rev?

/b

On Dec 8, 2008, at 3:36 PM, John Rutherford wrote:

> Any help would be greatly appreciated.  I have a pcap and the  
> freeSWITCH logs, and I can easily reproduce this.
>
> Thanks!
> John

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From john at loopfx.com  Mon Dec  8 14:16:42 2008
From: john at loopfx.com (John Rutherford)
Date: Mon, 8 Dec 2008 17:16:42 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D5A3@anniesue.loop.local>
	<244F3A58-EA28-4A76-AD5E-160E65B5B421@freeswitch.org>
Message-ID: <81469655CA61444CBB034826ABC6F6E331D5BD@anniesue.loop.local>

Sorry.  I forgot to mention that.  

 

I checked out the trunk last week.  I have revision 10597.  

 

John

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Monday, December 08, 2008 4:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

Are you on SVN trunk?  If not what rev?

 

/b

 

On Dec 8, 2008, at 3:36 PM, John Rutherford wrote:





Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this. 

 

Thanks!

John

 

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From jpalley at idapted.com  Mon Dec  8 20:37:52 2008
From: jpalley at idapted.com (Jonathan Palley)
Date: Tue, 9 Dec 2008 12:37:52 +0800
Subject: [Freeswitch-users] Jitter + Packet Loss
Message-ID: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>

I'm curious to start a discussion on being able to query a channel and get
statistics on the incoming jitter and packet loss (calculated from the RTP,
not RTCP).

Is this on the roadmap?  Is it hard to do?

Would be very useful for us indeed!

Thanks -
JP
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From regs at kinetix.gr  Tue Dec  9 00:21:08 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Tue, 09 Dec 2008 10:21:08 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>
	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
Message-ID: <493E2A74.7010502@kinetix.gr>

"I already added 2 patches for you right.  Just be clear about what you 
want."

And I am grateful of that.

"it is protocol neutral, that's why it starts with sip_"

I didn't know that. I thought that the sip_ variables are protocol 
specific. So one would expect there to be an iax_hangup_disposition,
woomera_hangup_disposition etc?

"Maybe you should beat around the bush less with your "requirements" for 
your application you are expecting me to support for you."

I am just trying to gather statistics for my providers as I would with 
any VoIP softswitch. (hangup causes per terminator per destination)
I don't think that this is a specific "application" rather than a 
general necessity for VoIP carriers. It is also very useful for 
troubleshooting
purposes : when I look at my CDRs to find a call that I got a complain 
for, I want to be able to tell if it was me or the provider who
hanged up and gave a specific hangup cause, so that I can troubleshoot 
the issue better.

"Just be clear about what you want."

I want FS to reach that level of detailing and maturity in all aspects 
so that it could be the softswitch of choice by any VoIP entrepreneur
(or hobbyist) and it is my strong belief that this can only be done by 
the community giving feedback to the programmers about what
they find useful or not (i.e. experience from real-life situations). The 
patches that you made the last few days *were not intended for
me* exclusively but for *anyone* that will face the same situations 
using FS. If you want the community to stop sending feedback about
features/improvements you may as well close down this mailing list or 
just use it as an announcement board.

I wish I was a c programmer and get involved with the project actively. 
But I am not. And as far as I can tell most of the registered users
in this list aren't either. So they only way we can help is by testing 
and suggesting.

Anthony Minessale wrote:
> it is protocol neutral, that's why it starts with sip_
>
> the variable can be any of:
>
> send_bye
> recv_bye
> send_cancel
> send_refuse
>
>
> using that value you can determine the information you asked.  I 
> answered your specific question which was:
> determining "which side hanged up".  Maybe you should beat around the 
> bush less with your "requirements" for your application you are 
> expecting me to support for you.
>
> I already added 2 patches for you right.  Just be clear about what you 
> want.
>
>
>
> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>     Not necessarily. For instance I got a "send_cancel" when the
>     calling party hanged up before the other party could pick up.
>     Also, shouldn't something like that be protocol/technology
>     neutral?
>
>
>
>     Anthony Minessale wrote:
>>     sip_hangup_disposition will be set to recv_bye on the side that
>>     was hungup.
>>
>>
>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>>
>>         Any updates about the "which side hanged up" potential variable?
>>
>>         Michael S Collins wrote:
>>>         Makes sense. I will look into this. 
>>>         -MC
>>>
>>>
>>>         On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos
>>>         <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>>>
>>>>         I am sending this second email to request/suggest/enquire
>>>>         about something relevant :
>>>>
>>>>         Wouldn't it be useful to know which end of a specific call
>>>>         leg send the protocol
>>>>         specific hangup cause? Otherwise it would be difficult to
>>>>         understand what really happened.
>>>>
>>>>
>>>>
>>>>         Michael S Collins wrote:
>>>>>         I will do some research on this and let you know what I find out.  
>>>>>         Question: are these internal calls or pstn or ?? Just curious about  
>>>>>         your environment.
>>>>>
>>>>>         Thanks,
>>>>>         MC
>>>>>
>>>>>
>>>>>
>>>>>         On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos < <mailto:regs at kinetix.gr>regs at kinetix.gr <mailto:regs at kinetix.gr>>  
>>>>>         wrote:
>>>>>
>>>>>           
>>>>>>         The proto_specific_hangup_cause is missing on one of the two
>>>>>>         call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>>         When the callee hangs up it is missing from the b-leg CDR. Is this  
>>>>>>         nornal?
>>>>>>
>>>>>>         And another question : what piece of info could inform me about who
>>>>>>         hanged up?
>>>>>>
>>>>>>
>>>>>>         -- 
>>>>>>         -------------------------------------------
>>>>>>         Apostolos Pantsiopoulos
>>>>>>         Kinetix Tele.com R & D
>>>>>>         email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>>         -------------------------------------------
>>>>>>
>>>>>>
>>>>>>         _______________________________________________
>>>>>>         Freeswitch-users mailing list
>>>>>>         Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>         http://www.freeswitch.org
>>>>>>             
>>>>>         _______________________________________________
>>>>>         Freeswitch-users mailing list
>>>>>         Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>         http://www.freeswitch.org
>>>>>           
>>>>
>>>>
>>>>         -- 
>>>>         -------------------------------------------
>>>>         Apostolos Pantsiopoulos
>>>>         Kinetix Tele.com R & D
>>>>         email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>         ------------------------------------------- 
>>>>         _______________________________________________
>>>>         Freeswitch-users mailing list
>>>>         Freeswitch-users at lists.freeswitch.org
>>>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org
>>>         ------------------------------------------------------------------------
>>>         _______________________________________________
>>>         Freeswitch-users mailing list
>>>         Freeswitch-users at lists.freeswitch.org
>>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>
>>
>>         -- 
>>         -------------------------------------------
>>         Apostolos Pantsiopoulos
>>         Kinetix Tele.com R & D
>>         email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>         ------------------------------------------- 
>>
>>
>>         _______________________________________________
>>         Freeswitch-users mailing list
>>         Freeswitch-users at lists.freeswitch.org
>>         <mailto:Freeswitch-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Anthony Minessale II
>>
>>     FreeSWITCH http://www.freeswitch.org/
>>     ClueCon http://www.cluecon.com/
>>
>>     AIM: anthm
>>     MSN:anthony_minessale at hotmail.com
>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>>     FreeSWITCH Developer Conference
>>     sip:888 at conference.freeswitch.org
>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>     iax:guest at conference.freeswitch.org/888
>>     <http://iax:guest at conference.freeswitch.org/888>
>>     googletalk:conf+888 at conference.freeswitch.org
>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>     pstn:213-799-1400
>>     ------------------------------------------------------------------------
>>     _______________________________________________ Freeswitch-users
>>     mailing list Freeswitch-users at lists.freeswitch.org
>>     <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From gkuri at ieee.org  Tue Dec  9 00:43:12 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Tue, 09 Dec 2008 00:43:12 -0800
Subject: [Freeswitch-users] root privs for mod_fax
Message-ID: <493E2FA0.5020006@ieee.org>

I've been experimenting with mod_fax and discovered it doesn't appear to
receive faxes unless freeswitch is running as root? it fails trying to
open the tiff file for writing (see the logs below). I'm using the
dialplan as prescribed in the wiki without any changes and the user the
freeswitch process is running under has privs to write to /tmp, but it
still fails to receive faxes. I haven't tried sending any faxes yet.

I'm running r10609.

any ideas?


2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Get document at 14400bps, modem 7
2008-12-09 00:29:41 [WARNING] mod_fax.c:133 spanfax_log_message()
WARNING T.30 Cannot open target TIFF file 'rxfax.tiff'
2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Changing from state 17 to 3
2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Tx:  DCN with final frame tag
2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Tx:  ff 13 fa
2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 HDLC carrier down in state 3
2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D
_TX
2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX
Set rx type 0
2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX
Set tx type 4
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Send complete in phase T30_PHASE_D_TX, state 3
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Send complete in phase T30_PHASE_D_TX, state 3
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Disconnecting
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX
Set rx type 0
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX
Set tx type 1
2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Changing from state 3 to 2
2008-12-09 00:29:44 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW
T.30 Send complete in phase T30_PHASE_E, state 2
2008-12-09 00:29:44 [DEBUG] mod_fax.c:163 phase_e_handler()
===============================================================
===============
2008-12-09 00:29:44 [DEBUG] mod_fax.c:176 phase_e_handler() Fax
processing not successful - result (41) TIFF/F file cannot
be opened.



Gabe



From mrjoebain at gmail.com  Tue Dec  9 01:27:46 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 9 Dec 2008 09:27:46 +0000
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
Message-ID: <748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>

On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
> Hi,
>
> I'm writing an IVR in Lua and am having problems dealing with hangups
> cleanly. Very often session:ready() reports true long after I have hung up
> and the hangup hook function I have set doesn't get called either. It
seems
> to report that the session is active indefinitely in some cases where a
loop
> keeps trying to get some dtmf key presses. Is there any trick to using
> session:ready() or the hangup hook that I might have missed?
>
> On a slightly related point I can't seem to access the session properties,
> e.g. session.caller_id_num has a value of nil. Any thoughts here?
>
> Thanks in advance,
>
> Joe Bain
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
2008/12/8 Michael Collins <msc at freeswitch.org>

> Joe,
>
> A few questions... what svn rev are you running? Which operating
> system? Finally, is it possible for you to put your dialplan and Lua
> script up at pastebin.freeswitch.org?
>
> Thanks,
> MC
>
Hi,

I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I can post
the dialplan and lua script though at the moment I can't seem to log in to
the pastebin, I just became a member on the freeswitch homepage but the
pass/username isn't being accepted.

Joe
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From ivan at myrvold.org  Tue Dec  9 01:40:19 2008
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 9 Dec 2008 10:40:19 +0100
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
Message-ID: <1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>

Did you read carefully when asked to provide login and password?
The login and password is there, don't use your own freeswitch login.

Ivan

Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:

> On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
> > Hi,
> >
> > I'm writing an IVR in Lua and am having problems dealing with  
> hangups
> > cleanly. Very often session:ready() reports true long after I have  
> hung up
> > and the hangup hook function I have set doesn't get called either.  
> It seems
> > to report that the session is active indefinitely in some cases  
> where a loop
> > keeps trying to get some dtmf key presses. Is there any trick to  
> using
> > session:ready() or the hangup hook that I might have missed?
> >
> > On a slightly related point I can't seem to access the session  
> properties,
> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
> >
> > Thanks in advance,
> >
> > Joe Bain
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 2008/12/8 Michael Collins <msc at freeswitch.org>
> Joe,
>
> A few questions... what svn rev are you running? Which operating
> system? Finally, is it possible for you to put your dialplan and Lua
> script up at pastebin.freeswitch.org?
>
> Thanks,
> MC
> Hi,
>
> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I  
> can post the dialplan and lua script though at the moment I can't  
> seem to log in to the pastebin, I just became a member on the  
> freeswitch homepage but the pass/username isn't being accepted.
>
> Joe
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mrjoebain at gmail.com  Tue Dec  9 02:06:10 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 9 Dec 2008 10:06:10 +0000
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
	<1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
Message-ID: <748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>

2008/12/9 Ivan C Myrvold <ivan at myrvold.org>

> Did you read carefully when asked to provide login and password? The login
> and password is there, don't use your own freeswitch login.
>
> Ivan
>
>  Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>
>  On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
> > Hi,
> >
> > I'm writing an IVR in Lua and am having problems dealing with hangups
> > cleanly. Very often session:ready() reports true long after I have hung
> up
> > and the hangup hook function I have set doesn't get called either. It
> seems
> > to report that the session is active indefinitely in some cases where a
> loop
> > keeps trying to get some dtmf key presses. Is there any trick to using
> > session:ready() or the hangup hook that I might have missed?
> >
> > On a slightly related point I can't seem to access the session
> properties,
> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
> >
> > Thanks in advance,
> >
> > Joe Bain
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 2008/12/8 Michael Collins <msc at freeswitch.org>
>
>> Joe,
>>
>> A few questions... what svn rev are you running? Which operating
>> system? Finally, is it possible for you to put your dialplan and Lua
>> script up at pastebin.freeswitch.org?
>>
>> Thanks,
>> MC
>>
> Hi,
>
> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I can post
> the dialplan and lua script though at the moment I can't seem to log in to
> the pastebin, I just became a member on the freeswitch homepage but the
> pass/username isn't being accepted.
>
> Joe
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
Ah, I should have read more carefully!

The dialplan is here <http://pastebin.freeswitch.org/6372> and the two
important lua scripts are here <http://pastebin.freeswitch.org/6371> and
here <http://pastebin.freeswitch.org/6370>, the first calls the second. I
didn't include all the Lua script as the problem appears right at the start
(as well as throughout) if the user hangs up when the IVR is speaking
(asking for an id number) then it seems to never get a hangup and loops
trying to get the id number.

Joe
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From helmut.kuper at ewetel.de  Tue Dec  9 02:07:27 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 09 Dec 2008 11:07:27 +0100
Subject: [Freeswitch-users] FS mod_fax
Message-ID: <493E435F.4010402@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I tried to compile mod_fax today with trunk from yesterday. A 'make' in
FS trunk directory led to an error saying that libspandsp.la wasn't
found in libs/spandsp/src. So I had to configure and compile (make)
spandsp manually before compiling FS.

regards
helmut

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BO8AoLa5wUOBqaEG1pOG4Qow8r7J2NF7
=MOUJ
-----END PGP SIGNATURE-----



From helmut.kuper at ewetel.de  Tue Dec  9 02:12:12 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 09 Dec 2008 11:12:12 +0100
Subject: [Freeswitch-users] mod_xml_ldap
Message-ID: <493E447C.5060507@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I tried compile FS with mod_xml_ldap with trunk of yesterday. During
compiling it can't find
http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I looked
there and found that the filename on freeswitch.org side has changed to
http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tar.gz


regards
helmut
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From jan.kubr at gmail.com  Tue Dec  9 03:52:24 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Tue, 9 Dec 2008 12:52:24 +0100
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
	<5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
	<191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
Message-ID: <698401620812090352x18ef62c2of7abceef9055fb4d@mail.gmail.com>

> btw you can send
>
> call-command: hangup
> hangup-cause: normal_clearing
>
> in place of
> call-command: execute
> execute-app-name: hangup
> execute-app-arg: normal_clearing

What is the difference this makes? Just curious because I've been
using the latter as well.


> we just tested you changes and it works the opposite way it should.
>
> this means: when we do not send an uuid, we get an an error
> (Reply-Text => -ERR invalid session id []). if we send a wrong/not
> existing uuid, the command will be executed on the inbound uuid.

This hasn't been changed, has it? On the latest trunk, if I don't pass
the uuid, I get "-ERR invalid session id". I can always pass it
explicitly though, so no big deal.


Jan Kubr



From jan.kubr at gmail.com  Tue Dec  9 03:53:13 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Tue, 9 Dec 2008 12:53:13 +0100
Subject: [Freeswitch-users] Read app ignores custom variable when called
	via socket interface
In-Reply-To: <191c3a030812080918x6acd7564q3247055cdc941641@mail.gmail.com>
References: <698401620812070331w7f8625ccv7e669f96f17f1e92@mail.gmail.com>
	<191c3a030812071144y1b0452c0k7bf529fd9604df96@mail.gmail.com>
	<698401620812080248u5f866d7es950d0019686b2c99@mail.gmail.com>
	<191c3a030812080918x6acd7564q3247055cdc941641@mail.gmail.com>
Message-ID: <698401620812090353m75feb135y4327eb42a0f7a4b@mail.gmail.com>

Updated and works great, thanks!

On Mon, Dec 8, 2008 at 6:18 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i added a patch to index the variables on the
> SWITCH_EVENT_CHANNEL_EXECUTE_COMPLETE
> if you want to update
>
> otherwise you can use uuid_getvar to retrieve the variable
>
>
> On Mon, Dec 8, 2008 at 4:48 AM, Jan Kubr <jan.kubr at gmail.com> wrote:
>>
>> OK my bad. The variable is set (I can see it in the Freeswitch console
>> when I use the info app), but they are only not send to me via the
>> socket interface. I get the "variable_*" variables only in the
>> beginning (after calling connect), but not in the events. How do I
>> enable this?
>> Thanks,
>> Jan
>>
>> On Sun, Dec 7, 2008 at 8:44 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > That seems unlikely.
>> > You sure about that?
>> >
>> > The var param is in the middle of the data which is passed as 1 giant
>> > string
>> > to the same exact app execution code.
>> > I don't see how it could differentiate
>> >
>> > did you try executing the info app right after to see all the vars.
>> >
>> > I'm not saying i don't believe you but it seems fishy.
>> >
>> >
>> >
>> > On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr <jan.kubr at gmail.com> wrote:
>> >>
>> >> Hi,
>> >> I checked out the current trunk (rev 10641) and found out that the
>> >> read app ignores the varname parameter, it always puts the result in
>> >> the DTMF-Digit variable. I'm calling it via the socket interface:
>> >>
>> >> sendmsg
>> >> call-command: execute
>> >> execute-app-name: read
>> >> execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res
>> >> 10000 #
>> >> event-lock:true
>> >>
>> >>
>> >> In the XML dialplan it works fine:
>> >>
>> >> <action application="read" data="1 1
>> >> sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
>> >> <action application="phrase" data="spell,${res}"/>
>> >>
>> >>
>> >> I have been using the socket call above successfully in the 1.0.1
>> >> release.
>> >>
>> >> Any ideas? Thanks,
>> >>
>> >> Jan Kubr
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From frank at impactfax.com  Tue Dec  9 05:31:33 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 9 Dec 2008 08:31:33 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <F8B619FF-4F53-4374-B903-871DB1C938E4@jerris.com>
Message-ID: <056d01c95a02$742fe4a0$33014c0a@ws4>

We are actually trying to detect the called party pressing a key - dtmf.
In band for ulaw.  Rfc2833 for 729.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Monday, December 08, 2008 12:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call
 
Are you really trying to detect a tone, or are you trying to detect dtmf
(could be delivered via rfc2833, info, etc) ?
 
Mike
 
On Dec 8, 2008, at 12:51 PM, Dan Le wrote:



>From my understanding, I didn't think tone_detect detects DTMF since
it's dual frequencies, rather tone_detect detects single frequencies
like fax tones. 
I would just run an IVR with a session.read or session.getDigits to
collect DTMF.
Dan
 
On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com>
wrote:
Same thing with version 10640 build.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Saturday, December 06, 2008 1:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call
make current or install current svn on a different box.

/b

On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:

>
> Ideas?  Am I doing something stupid or is tone_detect not just right
> here?
 
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From frank at impactfax.com  Tue Dec  9 05:38:27 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 9 Dec 2008 08:38:27 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
Message-ID: <057701c95a03$6ac5f610$33014c0a@ws4>

How can FS force a Minimum call duration for a FS caller (someone
calling out of FS)?  
 
We have a carrier that penalizes us with a surcharge for short duration
calls (sound familiar?).
 
So when a FS caller (not a call center or predictive dialer) calls a
cell phone and gets a ring tone or calls an answering machine, the FS
caller hangs up because they do not want to leave a message.  But they
do this in less then a few seconds after the call is answered.  This
becomes a short duration call and bang the surcharge applies.  It is
actually cheaper to pay for a longer call time (6 seconds in this case)
and avoid the short duration surcharge.  But the FS caller does not know
this.
 
So, how can FS hold the connection to the called party open for at least
the minimum amount of time I need to avoid the short call charge. even
though my FS caller has already hung up the phone on his end?  I would
like to do this in the xml dialplan if possible.
 
Thanks
 
-Frank
 
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From jaybinks at gmail.com  Tue Dec  9 05:53:13 2008
From: jaybinks at gmail.com (jay binks)
Date: Tue, 9 Dec 2008 23:53:13 +1000
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
Message-ID: <ec2ce17e0812090553r2f79ae39n4f90cf7dbc4aacd1@mail.gmail.com>

id also love to get any info from the RTCP...
even have this in the XML CDR would be great..
would love to derive quality stats for calls based on RTCP

Jay

On Tue, Dec 9, 2008 at 2:37 PM, Jonathan Palley <jpalley at idapted.com> wrote:

> I'm curious to start a discussion on being able to query a channel and get
> statistics on the incoming jitter and packet loss (calculated from the RTP,
> not RTCP).
>
> Is this on the roadmap?  Is it hard to do?
>
> Would be very useful for us indeed!
>
> Thanks -
> JP
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From brian at freeswitch.org  Tue Dec  9 06:01:59 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 08:01:59 -0600
Subject: [Freeswitch-users] root privs for mod_fax
In-Reply-To: <493E2FA0.5020006@ieee.org>
References: <493E2FA0.5020006@ieee.org>
Message-ID: <7BF70356-892B-45B2-B2D0-B5D3D1B7D01E@freeswitch.org>

If you're running SELinux then you'll need to correct that on your  
machine to allow FreeSWITCH to write to /tmp

/b

On Dec 9, 2008, at 2:43 AM, Gabriel Kuri wrote:

> I've been experimenting with mod_fax and discovered it doesn't  
> appear to
> receive faxes unless freeswitch is running as root? it fails trying to
> open the tiff file for writing (see the logs below). I'm using the
> dialplan as prescribed in the wiki without any changes and the user  
> the
> freeswitch process is running under has privs to write to /tmp, but it
> still fails to receive faxes. I haven't tried sending any faxes yet.
>
> I'm running r10609.
>
> any ideas?




From mrjoebain at gmail.com  Tue Dec  9 06:10:00 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 9 Dec 2008 14:10:00 +0000
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
	<1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
	<748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>
Message-ID: <748d53500812090610t70a11a07u594541a8e132a9d3@mail.gmail.com>

Ok I have been testing more and I have reduced my problem to a pretty short
and simple Lua script. I've posted it at
http://pastebin.freeswitch.org/6373 and
this gets called straight from the dialplan. From my experience so far it
only exits after a caller hangup about 1 in 10 times. Most of the time it
continues to loop until I do 'hupall'.

Thanks in advance if anyone can solve this or offer any advice.

Joe

2008/12/9 Joe Bain <mrjoebain at gmail.com>

> 2008/12/9 Ivan C Myrvold <ivan at myrvold.org>
>
>  Did you read carefully when asked to provide login and password? The
>> login and password is there, don't use your own freeswitch login.
>>
>> Ivan
>>
>>  Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>>
>>  On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
>> > Hi,
>> >
>> > I'm writing an IVR in Lua and am having problems dealing with hangups
>> > cleanly. Very often session:ready() reports true long after I have hung
>> up
>> > and the hangup hook function I have set doesn't get called either. It
>> seems
>> > to report that the session is active indefinitely in some cases where a
>> loop
>> > keeps trying to get some dtmf key presses. Is there any trick to using
>> > session:ready() or the hangup hook that I might have missed?
>> >
>> > On a slightly related point I can't seem to access the session
>> properties,
>> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
>> >
>> > Thanks in advance,
>> >
>> > Joe Bain
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 2008/12/8 Michael Collins <msc at freeswitch.org>
>>
>>> Joe,
>>>
>>> A few questions... what svn rev are you running? Which operating
>>> system? Finally, is it possible for you to put your dialplan and Lua
>>> script up at pastebin.freeswitch.org?
>>>
>>> Thanks,
>>> MC
>>>
>> Hi,
>>
>> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I can post
>> the dialplan and lua script though at the moment I can't seem to log in to
>> the pastebin, I just became a member on the freeswitch homepage but the
>> pass/username isn't being accepted.
>>
>> Joe
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> Ah, I should have read more carefully!
>
> The dialplan is here <http://pastebin.freeswitch.org/6372> and the two
> important lua scripts are here <http://pastebin.freeswitch.org/6371> and
> here <http://pastebin.freeswitch.org/6370>, the first calls the second. I
> didn't include all the Lua script as the problem appears right at the start
> (as well as throughout) if the user hangs up when the IVR is speaking
> (asking for an id number) then it seems to never get a hangup and loops
> trying to get the id number.
>
> Joe
>
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From erick at junctionnetworks.com  Mon Dec  8 17:14:00 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Mon, 08 Dec 2008 20:14:00 -0500
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
Message-ID: <493DC658.8020305@junctionnetworks.com>

Hi There,

I'm trying to get freeswitch to originate all SIP calls through
an outbound proxy.  When I use the originate API command to create a call
to a telephone number I see the SIP packets getting to my proxy just fine.
However if I originate a call to a SIP address then proxy server is 
bypassed,
instead FS is directly messaging the addressee.

Here is the command that I'm trying to use that behaves unexpectedly:
originate sofia/gateway/proxy/alice at bar.com &echo()

However this command produces the results I'm expecting:
originate sofia/gateway/proxy/15551234 &echo()

Here is the result of my sofia status:

freeswitch> sofia status
API CALL [sofia(status)] output:
                     Name          Type                               
Data      State
=================================================================================================
                 external       profile   
sip:mod_sofia at X.X.X.X:5070      RUNNING (0)
                    proxy       gateway   
sip:ejjohnson_ippx at ejjohnson.org      NOREG
=================================================================================================
1 profile 0 aliases

I have also tried setting the sip_invite_domain channel var through  the {}

Could you let me know what I'm doing wrong?

Much appreciated,

Erick J




From scott.ellis at novatex.com.au  Mon Dec  8 21:21:46 2008
From: scott.ellis at novatex.com.au (Scott Ellis)
Date: Tue, 09 Dec 2008 16:21:46 +1100
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
Message-ID: <493E006A.6030507@novatex.com.au>

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From brian at freeswitch.org  Tue Dec  9 06:30:03 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 08:30:03 -0600
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493DC658.8020305@junctionnetworks.com>
References: <493DC658.8020305@junctionnetworks.com>
Message-ID: <26B445AB-3DC6-4A45-B487-2DD7B67B4BA5@freeswitch.org>

First example is WRONG you don't dial via a gateway that way.  If you  
wish to dial alice at bar.com then try sofia/internal/alice at bar.com as  
you don't require a gateway to call alice right?

/b

On Dec 8, 2008, at 7:14 PM, Erick Johnson wrote:

> Here is the command that I'm trying to use that behaves unexpectedly:
> originate sofia/gateway/proxy/alice at bar.com &echo()
>
> However this command produces the results I'm expecting:
> originate sofia/gateway/proxy/15551234 &echo()

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From msc at freeswitch.org  Tue Dec  9 06:44:51 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 06:44:51 -0800
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493E2A74.7010502@kinetix.gr>
References: <49391D40.6050103@kinetix.gr>
	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>
	<49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
	<493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
	<493D2B9B.6050205@kinetix.gr>
	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
	<493E2A74.7010502@kinetix.gr>
Message-ID: <87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>

Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.

Do you have FS in production right now? Just curious.

Thanks,
MC

On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<regs at kinetix.gr> wrote:
> "I already added 2 patches for you right.  Just be clear about what you
> want."
>
> And I am grateful of that.
>
> "it is protocol neutral, that's why it starts with sip_"
>
> I didn't know that. I thought that the sip_ variables are protocol specific.
> So one would expect there to be an iax_hangup_disposition,
> woomera_hangup_disposition etc?
>
> "Maybe you should beat around the bush less with your "requirements" for
> your application you are expecting me to support for you."
>
> I am just trying to gather statistics for my providers as I would with any
> VoIP softswitch. (hangup causes per terminator per destination)
> I don't think that this is a specific "application" rather than a general
> necessity for VoIP carriers. It is also very useful for troubleshooting
> purposes : when I look at my CDRs to find a call that I got a complain for,
> I want to be able to tell if it was me or the provider who
> hanged up and gave a specific hangup cause, so that I can troubleshoot the
> issue better.
>
> "Just be clear about what you want."
>
> I want FS to reach that level of detailing and maturity in all aspects so
> that it could be the softswitch of choice by any VoIP entrepreneur
> (or hobbyist) and it is my strong belief that this can only be done by the
> community giving feedback to the programmers about what
> they find useful or not (i.e. experience from real-life situations). The
> patches that you made the last few days were not intended for
> me exclusively but for anyone that will face the same situations using FS.
> If you want the community to stop sending feedback about
> features/improvements you may as well close down this mailing list or just
> use it as an announcement board.
>
> I wish I was a c programmer and get involved with the project actively. But
> I am not. And as far as I can tell most of the registered users
> in this list aren't either. So they only way we can help is by testing and
> suggesting.
>
> Anthony Minessale wrote:
>
> it is protocol neutral, that's why it starts with sip_
>
> the variable can be any of:
>
> send_bye
> recv_bye
> send_cancel
> send_refuse
>
>
> using that value you can determine the information you asked.  I answered
> your specific question which was:
> determining "which side hanged up".  Maybe you should beat around the bush
> less with your "requirements" for your application you are expecting me to
> support for you.
>
> I already added 2 patches for you right.  Just be clear about what you want.
>
>
>
> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
> wrote:
>>
>> Not necessarily. For instance I got a "send_cancel" when the
>> calling party hanged up before the other party could pick up.
>> Also, shouldn't something like that be protocol/technology
>> neutral?
>>
>>
>>
>> Anthony Minessale wrote:
>>
>> sip_hangup_disposition will be set to recv_bye on the side that was
>> hungup.
>>
>>
>> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>> wrote:
>>>
>>> Any updates about the "which side hanged up" potential variable?
>>>
>>> Michael S Collins wrote:
>>>
>>> Makes sense. I will look into this.
>>> -MC
>>>
>>>
>>> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>> wrote:
>>>
>>> I am sending this second email to request/suggest/enquire about something
>>> relevant :
>>>
>>> Wouldn't it be useful to know which end of a specific call leg send the
>>> protocol
>>> specific hangup cause? Otherwise it would be difficult to understand what
>>> really happened.
>>>
>>>
>>>
>>> Michael S Collins wrote:
>>>
>>> I will do some research on this and let you know what I find out.
>>> Question: are these internal calls or pstn or ?? Just curious about
>>> your environment.
>>>
>>> Thanks,
>>> MC
>>>
>>>
>>>
>>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>> wrote:
>>>
>>>
>>>
>>> The proto_specific_hangup_cause is missing on one of the two
>>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>>> When the callee hangs up it is missing from the b-leg CDR. Is this
>>> nornal?
>>>
>>> And another question : what piece of info could inform me about who
>>> hanged up?
>>>
>>>
>>> --
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> ________________________________
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> ________________________________
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> ________________________________
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Tue Dec  9 06:50:07 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 06:50:07 -0800
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493E435F.4010402@ewetel.de>
References: <493E435F.4010402@ewetel.de>
Message-ID: <87f2f3b90812090650g67bd17a2w5a1e7490c1617abf@mail.gmail.com>

Which OS are you running?
-MC

On Tue, Dec 9, 2008 at 2:07 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I tried to compile mod_fax today with trunk from yesterday. A 'make' in
> FS trunk directory led to an error saying that libspandsp.la wasn't
> found in libs/spandsp/src. So I had to configure and compile (make)
> spandsp manually before compiling FS.
>
> regards
> helmut
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAkk+Q18ACgkQ4tZeNddg3dw5GgCgmmuLCsAx+T7IzUPayqAXZDaa
> BO8AoLa5wUOBqaEG1pOG4Qow8r7J2NF7
> =MOUJ
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From msc at freeswitch.org  Tue Dec  9 06:51:08 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 06:51:08 -0800
Subject: [Freeswitch-users] mod_xml_ldap
In-Reply-To: <493E447C.5060507@ewetel.de>
References: <493E447C.5060507@ewetel.de>
Message-ID: <87f2f3b90812090651p671d7a0fgd3f1bafd4f418f40@mail.gmail.com>

Thanks again for the heads up. We'll check it out.
-MC

On Tue, Dec 9, 2008 at 2:12 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I tried compile FS with mod_xml_ldap with trunk of yesterday. During
> compiling it can't find
> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I looked
> there and found that the filename on freeswitch.org side has changed to
> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tar.gz
>
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAkk+RHwACgkQ4tZeNddg3dyp7wCeN7fvIj4OSH1rsuglD46qtS36
> iR8AnjypwB2XT/rAYr61yyMXJ+iUY4/d
> =iJw/
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From gilbertandrew at me.com  Tue Dec  9 07:08:09 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 09 Dec 2008 10:08:09 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <057701c95a03$6ac5f610$33014c0a@ws4>
References: <057701c95a03$6ac5f610$33014c0a@ws4>
Message-ID: <4CF7C22E-64D4-4EEF-8153-4999E06E3A9F@me.com>

Don't want the tone to be wrong here, but this makes no sense.  
Carriers surcharge like this precisely to guard against call center,  
predictive and other mass outbound calling scenarios.

It just doesn't make since, math wise, that individuals hanging up on  
voice mail are going to significantly impact overall ACD stats, etc.  
So unless you have a very strange set of use cases or are pushing  
another category of traffic (ie call center) that skews you overall  
relationship with the carrier - I would go back and re-negotiate your  
arrangement.

Yes, FS is a b2bua and all is possible. But it is probably a better  
use of time to approach this as a business issue.

My 2 cents.


On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:

> How can FS force a Minimum call duration for a FS caller (someone  
> calling out of FS)?
>
> We have a carrier that penalizes us with a surcharge for short  
> duration calls (sound familiar?).
>
> So when a FS caller (not a call center or predictive dialer) calls a  
> cell phone and gets a ring tone or calls an answering machine, the  
> FS caller hangs up because they do not want to leave a message.  But  
> they do this in less then a few seconds after the call is answered.   
> This becomes a short duration call and bang the surcharge applies.   
> It is actually cheaper to pay for a longer call time (6 seconds in  
> this case) and avoid the short duration surcharge.  But the FS  
> caller does not know this.
>
> So, how can FS hold the connection to the called party open for at  
> least the minimum amount of time I need to avoid the short call  
> charge? even though my FS caller has already hung up the phone on  
> his end?  I would like to do this in the xml dialplanif possible.
>
> Thanks
>
> -Frank
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From regs at kinetix.gr  Tue Dec  9 07:19:26 2008
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Tue, 09 Dec 2008 17:19:26 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>	<70BE5500-DFEF-41A7-9928-B3369E45CED5@freeswitch.org>	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>
	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>
Message-ID: <493E8C7E.1050306@kinetix.gr>


    We are currently in the migration process from our
current system to a FS based setup. We are in the process of
adapting our billing and routing to FS. All the  CDRs (and variables)
related issues that we have been discussing on this mailing list
come from the need to extract the same level of information from FS as
we do with our current closed source proprietary system. So, we
chose FS because of the versatility it provides in every aspect (event
handling, config implementation etc.) and we strongly believe that all
these additions/fixes would be beneficial to many potential FS users.

    We are at your disposal for more details in case you need
more information about what exactly we are trying to do. Basically,
our approach is from the VoIP carrier's point of view rather than the
PBX user's/implementor's. So, the details that we asked to be introduced
to FS come from real life issues that we have faced during the last few 
years
with various platforms and troubleshooting experiences with other VoIP 
carriers.



Michael Collins wrote:
> Thanks for your feedback. It definitely helps to know not only what
> you need FS to do but why you need it to do so.
>
> Do you have FS in production right now? Just curious.
>
> Thanks,
> MC
>
> On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
> <regs at kinetix.gr> wrote:
>   
>> "I already added 2 patches for you right.  Just be clear about what you
>> want."
>>
>> And I am grateful of that.
>>
>> "it is protocol neutral, that's why it starts with sip_"
>>
>> I didn't know that. I thought that the sip_ variables are protocol specific.
>> So one would expect there to be an iax_hangup_disposition,
>> woomera_hangup_disposition etc?
>>
>> "Maybe you should beat around the bush less with your "requirements" for
>> your application you are expecting me to support for you."
>>
>> I am just trying to gather statistics for my providers as I would with any
>> VoIP softswitch. (hangup causes per terminator per destination)
>> I don't think that this is a specific "application" rather than a general
>> necessity for VoIP carriers. It is also very useful for troubleshooting
>> purposes : when I look at my CDRs to find a call that I got a complain for,
>> I want to be able to tell if it was me or the provider who
>> hanged up and gave a specific hangup cause, so that I can troubleshoot the
>> issue better.
>>
>> "Just be clear about what you want."
>>
>> I want FS to reach that level of detailing and maturity in all aspects so
>> that it could be the softswitch of choice by any VoIP entrepreneur
>> (or hobbyist) and it is my strong belief that this can only be done by the
>> community giving feedback to the programmers about what
>> they find useful or not (i.e. experience from real-life situations). The
>> patches that you made the last few days were not intended for
>> me exclusively but for anyone that will face the same situations using FS.
>> If you want the community to stop sending feedback about
>> features/improvements you may as well close down this mailing list or just
>> use it as an announcement board.
>>
>> I wish I was a c programmer and get involved with the project actively. But
>> I am not. And as far as I can tell most of the registered users
>> in this list aren't either. So they only way we can help is by testing and
>> suggesting.
>>
>> Anthony Minessale wrote:
>>
>> it is protocol neutral, that's why it starts with sip_
>>
>> the variable can be any of:
>>
>> send_bye
>> recv_bye
>> send_cancel
>> send_refuse
>>
>>
>> using that value you can determine the information you asked.  I answered
>> your specific question which was:
>> determining "which side hanged up".  Maybe you should beat around the bush
>> less with your "requirements" for your application you are expecting me to
>> support for you.
>>
>> I already added 2 patches for you right.  Just be clear about what you want.
>>
>>
>>
>> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>> wrote:
>>     
>>> Not necessarily. For instance I got a "send_cancel" when the
>>> calling party hanged up before the other party could pick up.
>>> Also, shouldn't something like that be protocol/technology
>>> neutral?
>>>
>>>
>>>
>>> Anthony Minessale wrote:
>>>
>>> sip_hangup_disposition will be set to recv_bye on the side that was
>>> hungup.
>>>
>>>
>>> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>> wrote:
>>>       
>>>> Any updates about the "which side hanged up" potential variable?
>>>>
>>>> Michael S Collins wrote:
>>>>
>>>> Makes sense. I will look into this.
>>>> -MC
>>>>
>>>>
>>>> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>>> wrote:
>>>>
>>>> I am sending this second email to request/suggest/enquire about something
>>>> relevant :
>>>>
>>>> Wouldn't it be useful to know which end of a specific call leg send the
>>>> protocol
>>>> specific hangup cause? Otherwise it would be difficult to understand what
>>>> really happened.
>>>>
>>>>
>>>>
>>>> Michael S Collins wrote:
>>>>
>>>> I will do some research on this and let you know what I find out.
>>>> Question: are these internal calls or pstn or ?? Just curious about
>>>> your environment.
>>>>
>>>> Thanks,
>>>> MC
>>>>
>>>>
>>>>
>>>> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>
>>>> wrote:
>>>>
>>>>
>>>>
>>>> The proto_specific_hangup_cause is missing on one of the two
>>>> call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>> When the callee hangs up it is missing from the b-leg CDR. Is this
>>>> nornal?
>>>>
>>>> And another question : what piece of info could inform me about who
>>>> hanged up?
>>>>
>>>>
>>>> --
>>>> -------------------------------------------
>>>> Apostolos Pantsiopoulos
>>>> Kinetix Tele.com R & D
>>>> email: regs at kinetix.gr
>>>> -------------------------------------------
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> -------------------------------------------
>>>> Apostolos Pantsiopoulos
>>>> Kinetix Tele.com R & D
>>>> email: regs at kinetix.gr
>>>> -------------------------------------------
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> ________________________________
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> -------------------------------------------
>>>> Apostolos Pantsiopoulos
>>>> Kinetix Tele.com R & D
>>>> email: regs at kinetix.gr
>>>> -------------------------------------------
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>         
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:213-799-1400
>>>
>>> ________________________________
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> -------------------------------------------
>>> Apostolos Pantsiopoulos
>>> Kinetix Tele.com R & D
>>> email: regs at kinetix.gr
>>> -------------------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> ________________________________
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> -------------------------------------------
>> Apostolos Pantsiopoulos
>> Kinetix Tele.com R & D
>> email: regs at kinetix.gr
>> -------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
------------------------------------------- 

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From mike at jerris.com  Tue Dec  9 07:45:56 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 10:45:56 -0500
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
Message-ID: <B7BC254D-855D-4247-82B9-BEF52C024E9E@jerris.com>

It is something we have been discussing as we need these stats to do  
rtcp properly but we have not written any code to do so.  It is  
"somewhat" difficult.  I would say it is on our minds but not on any  
roadmap just yet.

MIke

On Dec 8, 2008, at 11:37 PM, Jonathan Palley wrote:

> I'm curious to start a discussion on being able to query a channel  
> and get statistics on the incoming jitter and packet loss  
> (calculated from the RTP, not RTCP).
>
> Is this on the roadmap?  Is it hard to do?




From mike at jerris.com  Tue Dec  9 07:56:32 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 10:56:32 -0500
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493E435F.4010402@ewetel.de>
References: <493E435F.4010402@ewetel.de>
Message-ID: <FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>

make sure you have libtiff and libtiff dev packages installed then re- 
configure freeswitch

Mike

On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I tried to compile mod_fax today with trunk from yesterday. A 'make'  
> in
> FS trunk directory led to an error saying that libspandsp.la wasn't
> found in libs/spandsp/src. So I had to configure and compile (make)
> spandsp manually before compiling FS.
>
> regards
> helmut
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAkk+Q18ACgkQ4tZeNddg3dw5GgCgmmuLCsAx+T7IzUPayqAXZDaa
> BO8AoLa5wUOBqaEG1pOG4Qow8r7J2NF7
> =MOUJ
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Tue Dec  9 07:59:50 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 10:59:50 -0500
Subject: [Freeswitch-users] mod_xml_ldap
In-Reply-To: <87f2f3b90812090651p671d7a0fgd3f1bafd4f418f40@mail.gmail.com>
References: <493E447C.5060507@ewetel.de>
	<87f2f3b90812090651p671d7a0fgd3f1bafd4f418f40@mail.gmail.com>
Message-ID: <89D6643E-06C5-45B2-8244-D211C857B182@jerris.com>

Fixed in svn r10678.  Thanks for the report.

Mike

On Dec 9, 2008, at 9:51 AM, Michael Collins wrote:

> Thanks again for the heads up. We'll check it out.
> -MC
>
> On Tue, Dec 9, 2008 at 2:12 AM, Helmut Kuper  
> <helmut.kuper at ewetel.de> wrote:
>> I tried compile FS with mod_xml_ldap with trunk of yesterday. During
>> compiling it can't find
>> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I  
>> looked
>> there and found that the filename on freeswitch.org side has  
>> changed to
>> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tar.gz



From mike at jerris.com  Tue Dec  9 08:01:51 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 11:01:51 -0500
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <748d53500812090610t70a11a07u594541a8e132a9d3@mail.gmail.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
	<1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
	<748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>
	<748d53500812090610t70a11a07u594541a8e132a9d3@mail.gmail.com>
Message-ID: <8F96510A-1A1E-45E9-A206-FDA66CAEA06F@jerris.com>


On Dec 9, 2008, at 9:10 AM, Joe Bain wrote:

> Ok I have been testing more and I have reduced my problem to a  
> pretty short and simple Lua script. I've posted it at http://pastebin.freeswitch.org/6373 
>  and this gets called straight from the dialplan. From my experience  
> so far it only exits after a caller hangup about 1 in 10 times. Most  
> of the time it continues to loop until I do 'hupall'.
>
> Thanks in advance if anyone can solve this or offer any advice.
>
> Joe
>
> 2008/12/9 Joe Bain <mrjoebain at gmail.com>
> 2008/12/9 Ivan C Myrvold <ivan at myrvold.org>
>
> Did you read carefully when asked to provide login and password?
> The login and password is there, don't use your own freeswitch login.
>
> Ivan
>
> Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>
>> On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
>> > Hi,
>> >
>> > I'm writing an IVR in Lua and am having problems dealing with  
>> hangups
>> > cleanly. Very often session:ready() reports true long after I  
>> have hung up
>> > and the hangup hook function I have set doesn't get called  
>> either. It seems
>> > to report that the session is active indefinitely in some cases  
>> where a loop
>> > keeps trying to get some dtmf key presses. Is there any trick to  
>> using
>> > session:ready() or the hangup hook that I might have missed?
>> >
>> > On a slightly related point I can't seem to access the session  
>> properties,
>> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
>> >

>> Joe,
>>
>> A few questions... what svn rev are you running? Which operating
>> system? Finally, is it possible for you to put your dialplan and Lua
>> script up at pastebin.freeswitch.org?
>>
>> Thanks,
>> MC
>> Hi,
>>
>> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I  
>> can post the dialplan and lua script though at the moment I can't  
>> seem to log in to the pastebin, I just became a member on the  
>> freeswitch homepage but the pass/username isn't being accepted.
>>
>> Joe
>
>
> Ah, I should have read more carefully!
>
> The dialplan is here and the two important lua scripts are here and  
> here, the first calls the second. I didn't include all the Lua  
> script as the problem appears right at the start (as well as  
> throughout) if the user hangs up when the IVR is speaking (asking  
> for an id number) then it seems to never get a hangup and loops  
> trying to get the id number.
>
> Joe

We just tested this with current svn trunk and it appears to work  
fine, could you try updating and see if it is still a problem for you

Mike

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From mike at jerris.com  Tue Dec  9 08:06:02 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 11:06:02 -0500
Subject: [Freeswitch-users] key tone trigger event during call
In-Reply-To: <056d01c95a02$742fe4a0$33014c0a@ws4>
References: <056d01c95a02$742fe4a0$33014c0a@ws4>
Message-ID: <4B77D066-6FA0-4689-B9A7-B47B4429403D@jerris.com>

you don't want to be using the tone detect here, you want to be using  
bind_meta, but without the meta key, which I don't think it can  
actually do currently.

Mike

On Dec 9, 2008, at 8:31 AM, Frank @ Impact wrote:

> We are actually trying to detect the called party pressing a key ?  
> dtmf.  In band for ulaw.  Rfc2833 for 729.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Jerris
> Sent: Monday, December 08, 2008 12:57 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
>
> Are you really trying to detect a tone, or are you trying to detect  
> dtmf (could be delivered via rfc2833, info, etc) ?
>
> Mike
>
> On Dec 8, 2008, at 12:51 PM, Dan Le wrote:
>
>
> From my understanding, I didn't think tone_detect detects DTMF since  
> it's dual frequencies, rather tone_detect detects single frequencies  
> like fax tones.
>
> I would just run an IVR with a session.read or session.getDigits to  
> collect DTMF.
>
> Dan
>
>
> On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <frank at impactfax.com>  
> wrote:
> Same thing with version 10640 build.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian West
> Sent: Saturday, December 06, 2008 1:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
> make current or install current svn on a different box.
>
> /b
>
> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>
> >
> > Ideas?  Am I doing something stupid or is tone_detect not just right
> > here?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From helmut.kuper at ewetel.de  Tue Dec  9 08:09:10 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 09 Dec 2008 17:09:10 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>
References: <493E435F.4010402@ewetel.de>
	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>
Message-ID: <493E9826.108@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

don't know if you get me right: Everything is there, but obviously FS
makefile has to compile "libs/spandsp/src" before mod_fax (at least I
guess so). Currently the Makefile referred to libspandsp.la before it is
compiled.

regards
helmut



Michael Jerris schrieb:
> make sure you have libtiff and libtiff dev packages installed then re- 
> configure freeswitch
> 
> Mike
> 
> On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:
> 
> Hello,
> 
> I tried to compile mod_fax today with trunk from yesterday. A 'make'  
> in
> FS trunk directory led to an error saying that libspandsp.la wasn't
> found in libs/spandsp/src. So I had to configure and compile (make)
> spandsp manually before compiling FS.
> 
> regards
> helmut
> 
>>
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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=4Tja
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From mike at jerris.com  Tue Dec  9 08:10:06 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 11:10:06 -0500
Subject: [Freeswitch-users] Two major flaws: Could they be fixed?
In-Reply-To: <698401620812090352x18ef62c2of7abceef9055fb4d@mail.gmail.com>
References: <5e414ed0812080128t6c839f41i7bc2bc8d22c7ed82@mail.gmail.com>
	<191c3a030812080604q12aab032me2c2b5fe33ef31e0@mail.gmail.com>
	<5e414ed0812080747r44d410f5rb69ed8d6250e39a7@mail.gmail.com>
	<191c3a030812080808i6293c1cdm2ff16caaf1790b3f@mail.gmail.com>
	<5e414ed0812080819q2669a253i4bfa34aa4214a656@mail.gmail.com>
	<191c3a030812080844h65af77es3d1ae9b8753da42e@mail.gmail.com>
	<5e414ed0812080856j44550156v607754ea601381ab@mail.gmail.com>
	<191c3a030812080911g68c841b8ye778fec02825a770@mail.gmail.com>
	<698401620812090352x18ef62c2of7abceef9055fb4d@mail.gmail.com>
Message-ID: <4CBD44BD-3824-4BFD-BAC1-6E1DAE0C71E7@jerris.com>


On Dec 9, 2008, at 6:52 AM, Jan Kubr wrote:

>> btw you can send
>>
>> call-command: hangup
>> hangup-cause: normal_clearing
>>
>> in place of
>> call-command: execute
>> execute-app-name: hangup
>> execute-app-arg: normal_clearing
>
> What is the difference this makes? Just curious because I've been
> using the latter as well.
>
>
>> we just tested you changes and it works the opposite way it should.
>>
>> this means: when we do not send an uuid, we get an an error
>> (Reply-Text => -ERR invalid session id []). if we send a wrong/not
>> existing uuid, the command will be executed on the inbound uuid.
>
> This hasn't been changed, has it? On the latest trunk, if I don't pass
> the uuid, I get "-ERR invalid session id". I can always pass it
> explicitly though, so no big deal.
>
>
> Jan Kubr

We did have confirmation from others that this is working properly  
now.  Can you please make sure you are on current trunk and re-test  
this.

Mike




From sicfslist at gmail.com  Tue Dec  9 08:12:01 2008
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Tue, 9 Dec 2008 10:12:01 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493E8C7E.1050306@kinetix.gr>
References: <49391D40.6050103@kinetix.gr> <49395403.6080404@kinetix.gr>
	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>
	<493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
	<493D2B9B.6050205@kinetix.gr>
	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
	<493E2A74.7010502@kinetix.gr>
	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>
	<493E8C7E.1050306@kinetix.gr>
Message-ID: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>

Hello,
This is just my 2 cents ... but my experience has been that trying to catch
all of the various variables (i.e. from XML_CDR) or otherwise can be a
little trying (a row in your CDR database could be over 100 fields long!).

The best option here is to catch the UUID's for the 2 call legs, capture all
SIP messaging, parse and dump the messaging, and then correlate the calls
from the CDR from there.

Much easier than trying to do it from FS ... and most folks want to see SIP
captures anyway (very broad set of tools to debug).

Measuring things like ASR, PDD, etc in my opinion is much easier from the
raw messaging than trying to do something with FS CDR records.



On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs at kinetix.gr>wrote:

>
>     We are currently in the migration process from our
> current system to a FS based setup. We are in the process of
> adapting our billing and routing to FS. All the  CDRs (and variables)
> related issues that we have been discussing on this mailing list
> come from the need to extract the same level of information from FS as
> we do with our current closed source proprietary system. So, we
> chose FS because of the versatility it provides in every aspect (event
> handling, config implementation etc.) and we strongly believe that all
> these additions/fixes would be beneficial to many potential FS users.
>
>     We are at your disposal for more details in case you need
> more information about what exactly we are trying to do. Basically,
> our approach is from the VoIP carrier's point of view rather than the
> PBX user's/implementor's. So, the details that we asked to be introduced
> to FS come from real life issues that we have faced during the last few
> years
> with various platforms and troubleshooting experiences with other VoIP
> carriers.
>
>
>
>
> Michael Collins wrote:
>
> Thanks for your feedback. It definitely helps to know not only what
> you need FS to do but why you need it to do so.
>
> Do you have FS in production right now? Just curious.
>
> Thanks,
> MC
>
> On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos<regs at kinetix.gr> <regs at kinetix.gr> wrote:
>
>
>  "I already added 2 patches for you right.  Just be clear about what you
> want."
>
> And I am grateful of that.
>
> "it is protocol neutral, that's why it starts with sip_"
>
> I didn't know that. I thought that the sip_ variables are protocol specific.
> So one would expect there to be an iax_hangup_disposition,
> woomera_hangup_disposition etc?
>
> "Maybe you should beat around the bush less with your "requirements" for
> your application you are expecting me to support for you."
>
> I am just trying to gather statistics for my providers as I would with any
> VoIP softswitch. (hangup causes per terminator per destination)
> I don't think that this is a specific "application" rather than a general
> necessity for VoIP carriers. It is also very useful for troubleshooting
> purposes : when I look at my CDRs to find a call that I got a complain for,
> I want to be able to tell if it was me or the provider who
> hanged up and gave a specific hangup cause, so that I can troubleshoot the
> issue better.
>
> "Just be clear about what you want."
>
> I want FS to reach that level of detailing and maturity in all aspects so
> that it could be the softswitch of choice by any VoIP entrepreneur
> (or hobbyist) and it is my strong belief that this can only be done by the
> community giving feedback to the programmers about what
> they find useful or not (i.e. experience from real-life situations). The
> patches that you made the last few days were not intended for
> me exclusively but for anyone that will face the same situations using FS.
> If you want the community to stop sending feedback about
> features/improvements you may as well close down this mailing list or just
> use it as an announcement board.
>
> I wish I was a c programmer and get involved with the project actively. But
> I am not. And as far as I can tell most of the registered users
> in this list aren't either. So they only way we can help is by testing and
> suggesting.
>
> Anthony Minessale wrote:
>
> it is protocol neutral, that's why it starts with sip_
>
> the variable can be any of:
>
> send_bye
> recv_bye
> send_cancel
> send_refuse
>
>
> using that value you can determine the information you asked.  I answered
> your specific question which was:
> determining "which side hanged up".  Maybe you should beat around the bush
> less with your "requirements" for your application you are expecting me to
> support for you.
>
> I already added 2 patches for you right.  Just be clear about what you want.
>
>
>
> On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <regs at kinetix.gr>
> wrote:
>
>
>  Not necessarily. For instance I got a "send_cancel" when the
> calling party hanged up before the other party could pick up.
> Also, shouldn't something like that be protocol/technology
> neutral?
>
>
>
> Anthony Minessale wrote:
>
> sip_hangup_disposition will be set to recv_bye on the side that was
> hungup.
>
>
> On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <regs at kinetix.gr>
> wrote:
>
>
>  Any updates about the "which side hanged up" potential variable?
>
> Michael S Collins wrote:
>
> Makes sense. I will look into this.
> -MC
>
>
> On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <regs at kinetix.gr>
> wrote:
>
> I am sending this second email to request/suggest/enquire about something
> relevant :
>
> Wouldn't it be useful to know which end of a specific call leg send the
> protocol
> specific hangup cause? Otherwise it would be difficult to understand what
> really happened.
>
>
>
> Michael S Collins wrote:
>
> I will do some research on this and let you know what I find out.
> Question: are these internal calls or pstn or ?? Just curious about
> your environment.
>
> Thanks,
> MC
>
>
>
> On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <regs at kinetix.gr>
> wrote:
>
>
>
> The proto_specific_hangup_cause is missing on one of the two
> call legs. When the caller hangs up it is missing from the a-leg CDR.
> When the callee hangs up it is missing from the b-leg CDR. Is this
> nornal?
>
> And another question : what piece of info could inform me about who
> hanged up?
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
> ________________________________
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>          --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthmMSN:anthony_minessale at hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conferencesip:888 at conference.freeswitch.orgiax:guest at conference.freeswitch.org/888googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> ________________________________
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>        --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthmMSN:anthony_minessale at hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conferencesip:888 at conference.freeswitch.orgiax:guest at conference.freeswitch.org/888googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> ________________________________
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>      _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> -------------------------------------------
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: regs at kinetix.gr
> -------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Dec  9 08:25:22 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Dec 2008 11:25:22 -0500
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493E9826.108@ewetel.de>
References: <493E435F.4010402@ewetel.de>
	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>
	<493E9826.108@ewetel.de>
Message-ID: <724B6B1A-4834-4907-A9E8-73076981176C@jerris.com>


On Dec 9, 2008, at 11:09 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> don't know if you get me right: Everything is there, but obviously FS
> makefile has to compile "libs/spandsp/src" before mod_fax (at least I
> guess so). Currently the Makefile referred to libspandsp.la before  
> it is
> compiled.
>
> regards
> helmut

Did you try it?

Mike




From gkuri at ieee.org  Tue Dec  9 08:35:25 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Tue, 09 Dec 2008 08:35:25 -0800
Subject: [Freeswitch-users] root privs for mod_fax
In-Reply-To: <7BF70356-892B-45B2-B2D0-B5D3D1B7D01E@freeswitch.org>
References: <493E2FA0.5020006@ieee.org>
	<7BF70356-892B-45B2-B2D0-B5D3D1B7D01E@freeswitch.org>
Message-ID: <493E9E4D.7040809@ieee.org>

I'm running Gentoo Linux.

# uname -a

Linux 2.6.25-gentoo-r7 #1 SMP PREEMPT Sun Oct 5 01:51:24 PDT 2008 x86_64
Intel(R) Xeon(R) CPU X3320 @ 2.50GHz GenuineIntel GNU/Linux

/tmp is writable by everyone ...

# ls -ld /tmp
drwxrwxrwt 4 root root 4096 Dec  9 08:28 /tmp

ideas?

also, I assume the spool directory is just where it's spooling the file
temporarily while the fax is coming in, but is there a variable I can
set to tell it where to put the file after the fax has been received? it
looks like the time I ran FS as root and received a fax successfully,
the tiff file ended up in /root ?

Gabe

Brian West wrote:
> If you're running SELinux then you'll need to correct that on your  
> machine to allow FreeSWITCH to write to /tmp
> 
> /b
> 
> On Dec 9, 2008, at 2:43 AM, Gabriel Kuri wrote:
> 
>> I've been experimenting with mod_fax and discovered it doesn't  
>> appear to
>> receive faxes unless freeswitch is running as root? it fails trying to
>> open the tiff file for writing (see the logs below). I'm using the
>> dialplan as prescribed in the wiki without any changes and the user  
>> the
>> freeswitch process is running under has privs to write to /tmp, but it
>> still fails to receive faxes. I haven't tried sending any faxes yet.
>>
>> I'm running r10609.
>>
>> any ideas?
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 






From regs at kinetix.gr  Tue Dec  9 08:37:25 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Tue, 09 Dec 2008 18:37:25 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>
	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
Message-ID: <493E9EC5.1060701@kinetix.gr>

That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).

In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.



Shelby Ramsey wrote:
> Hello,
>
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise 
> can be a little trying (a row in your CDR database could be over 100 
> fields long!).  
>
> The best option here is to catch the UUID's for the 2 call legs, 
> capture all SIP messaging, parse and dump the messaging, and then 
> correlate the calls from the CDR from there.  
>
> Much easier than trying to do it from FS ... and most folks want to 
> see SIP captures anyway (very broad set of tools to debug).  
>
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.
>
>
>
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
>
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be
>     introduced
>     to FS come from real life issues that we have faced during the
>     last few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
>
>
>
>
>     Michael Collins wrote:
>>     Thanks for your feedback. It definitely helps to know not only what
>>     you need FS to do but why you need it to do so.
>>
>>     Do you have FS in production right now? Just curious.
>>
>>     Thanks,
>>     MC
>>
>>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "I already added 2 patches for you right.  Just be clear about what you
>>>     want."
>>>
>>>     And I am grateful of that.
>>>
>>>     "it is protocol neutral, that's why it starts with sip_"
>>>
>>>     I didn't know that. I thought that the sip_ variables are protocol specific.
>>>     So one would expect there to be an iax_hangup_disposition,
>>>     woomera_hangup_disposition etc?
>>>
>>>     "Maybe you should beat around the bush less with your "requirements" for
>>>     your application you are expecting me to support for you."
>>>
>>>     I am just trying to gather statistics for my providers as I would with any
>>>     VoIP softswitch. (hangup causes per terminator per destination)
>>>     I don't think that this is a specific "application" rather than a general
>>>     necessity for VoIP carriers. It is also very useful for troubleshooting
>>>     purposes : when I look at my CDRs to find a call that I got a complain for,
>>>     I want to be able to tell if it was me or the provider who
>>>     hanged up and gave a specific hangup cause, so that I can troubleshoot the
>>>     issue better.
>>>
>>>     "Just be clear about what you want."
>>>
>>>     I want FS to reach that level of detailing and maturity in all aspects so
>>>     that it could be the softswitch of choice by any VoIP entrepreneur
>>>     (or hobbyist) and it is my strong belief that this can only be done by the
>>>     community giving feedback to the programmers about what
>>>     they find useful or not (i.e. experience from real-life situations). The
>>>     patches that you made the last few days were not intended for
>>>     me exclusively but for anyone that will face the same situations using FS.
>>>     If you want the community to stop sending feedback about
>>>     features/improvements you may as well close down this mailing list or just
>>>     use it as an announcement board.
>>>
>>>     I wish I was a c programmer and get involved with the project actively. But
>>>     I am not. And as far as I can tell most of the registered users
>>>     in this list aren't either. So they only way we can help is by testing and
>>>     suggesting.
>>>
>>>     Anthony Minessale wrote:
>>>
>>>     it is protocol neutral, that's why it starts with sip_
>>>
>>>     the variable can be any of:
>>>
>>>     send_bye
>>>     recv_bye
>>>     send_cancel
>>>     send_refuse
>>>
>>>
>>>     using that value you can determine the information you asked.  I answered
>>>     your specific question which was:
>>>     determining "which side hanged up".  Maybe you should beat around the bush
>>>     less with your "requirements" for your application you are expecting me to
>>>     support for you.
>>>
>>>     I already added 2 patches for you right.  Just be clear about what you want.
>>>
>>>
>>>
>>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     Not necessarily. For instance I got a "send_cancel" when the
>>>>     calling party hanged up before the other party could pick up.
>>>>     Also, shouldn't something like that be protocol/technology
>>>>     neutral?
>>>>
>>>>
>>>>
>>>>     Anthony Minessale wrote:
>>>>
>>>>     sip_hangup_disposition will be set to recv_bye on the side that was
>>>>     hungup.
>>>>
>>>>
>>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     Any updates about the "which side hanged up" potential variable?
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     Makes sense. I will look into this.
>>>>>     -MC
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>     I am sending this second email to request/suggest/enquire about something
>>>>>     relevant :
>>>>>
>>>>>     Wouldn't it be useful to know which end of a specific call leg send the
>>>>>     protocol
>>>>>     specific hangup cause? Otherwise it would be difficult to understand what
>>>>>     really happened.
>>>>>
>>>>>
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     I will do some research on this and let you know what I find out.
>>>>>     Question: are these internal calls or pstn or ?? Just curious about
>>>>>     your environment.
>>>>>
>>>>>     Thanks,
>>>>>     MC
>>>>>
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>
>>>>>
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>     ________________________________
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>         
>>     _______________________________________________
>>     Freeswitch-users mailing list
>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From msc at freeswitch.org  Tue Dec  9 08:37:55 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 08:37:55 -0800
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493E9826.108@ewetel.de>
References: <493E435F.4010402@ewetel.de>
	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>
	<493E9826.108@ewetel.de>
Message-ID: <87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>

Helmut,
I think Mike J was pointing out that spandsp needs libtiff and
libtiff-devel in order to compile, so you need to do that first and
then compile freeswitch.
-MC

On Tue, Dec 9, 2008 at 8:09 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> don't know if you get me right: Everything is there, but obviously FS
> makefile has to compile "libs/spandsp/src" before mod_fax (at least I
> guess so). Currently the Makefile referred to libspandsp.la before it is
> compiled.
>
> regards
> helmut
>
>
>
> Michael Jerris schrieb:
>> make sure you have libtiff and libtiff dev packages installed then re-
>> configure freeswitch
>>
>> Mike
>>
>> On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:
>>
>> Hello,
>>
>> I tried to compile mod_fax today with trunk from yesterday. A 'make'
>> in
>> FS trunk directory led to an error saying that libspandsp.la wasn't
>> found in libs/spandsp/src. So I had to configure and compile (make)
>> spandsp manually before compiling FS.
>>
>> regards
>> helmut
>>
>>>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAkk+mCYACgkQ4tZeNddg3dxlPgCgpey84xCtTAD0GyiyDP3uPxpz
> SPgAnRJNO1s3n3xabGSbJYPtQmti2VKT
> =4Tja
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From regs at kinetix.gr  Tue Dec  9 08:38:21 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Tue, 09 Dec 2008 18:38:21 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>
	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
Message-ID: <493E9EFD.6040203@kinetix.gr>

That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).

In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.



Shelby Ramsey wrote:
> Hello,
>
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise 
> can be a little trying (a row in your CDR database could be over 100 
> fields long!).  
>
> The best option here is to catch the UUID's for the 2 call legs, 
> capture all SIP messaging, parse and dump the messaging, and then 
> correlate the calls from the CDR from there.  
>
> Much easier than trying to do it from FS ... and most folks want to 
> see SIP captures anyway (very broad set of tools to debug).  
>
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.
>
>
>
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
>
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be
>     introduced
>     to FS come from real life issues that we have faced during the
>     last few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
>
>
>
>
>     Michael Collins wrote:
>>     Thanks for your feedback. It definitely helps to know not only what
>>     you need FS to do but why you need it to do so.
>>
>>     Do you have FS in production right now? Just curious.
>>
>>     Thanks,
>>     MC
>>
>>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "I already added 2 patches for you right.  Just be clear about what you
>>>     want."
>>>
>>>     And I am grateful of that.
>>>
>>>     "it is protocol neutral, that's why it starts with sip_"
>>>
>>>     I didn't know that. I thought that the sip_ variables are protocol specific.
>>>     So one would expect there to be an iax_hangup_disposition,
>>>     woomera_hangup_disposition etc?
>>>
>>>     "Maybe you should beat around the bush less with your "requirements" for
>>>     your application you are expecting me to support for you."
>>>
>>>     I am just trying to gather statistics for my providers as I would with any
>>>     VoIP softswitch. (hangup causes per terminator per destination)
>>>     I don't think that this is a specific "application" rather than a general
>>>     necessity for VoIP carriers. It is also very useful for troubleshooting
>>>     purposes : when I look at my CDRs to find a call that I got a complain for,
>>>     I want to be able to tell if it was me or the provider who
>>>     hanged up and gave a specific hangup cause, so that I can troubleshoot the
>>>     issue better.
>>>
>>>     "Just be clear about what you want."
>>>
>>>     I want FS to reach that level of detailing and maturity in all aspects so
>>>     that it could be the softswitch of choice by any VoIP entrepreneur
>>>     (or hobbyist) and it is my strong belief that this can only be done by the
>>>     community giving feedback to the programmers about what
>>>     they find useful or not (i.e. experience from real-life situations). The
>>>     patches that you made the last few days were not intended for
>>>     me exclusively but for anyone that will face the same situations using FS.
>>>     If you want the community to stop sending feedback about
>>>     features/improvements you may as well close down this mailing list or just
>>>     use it as an announcement board.
>>>
>>>     I wish I was a c programmer and get involved with the project actively. But
>>>     I am not. And as far as I can tell most of the registered users
>>>     in this list aren't either. So they only way we can help is by testing and
>>>     suggesting.
>>>
>>>     Anthony Minessale wrote:
>>>
>>>     it is protocol neutral, that's why it starts with sip_
>>>
>>>     the variable can be any of:
>>>
>>>     send_bye
>>>     recv_bye
>>>     send_cancel
>>>     send_refuse
>>>
>>>
>>>     using that value you can determine the information you asked.  I answered
>>>     your specific question which was:
>>>     determining "which side hanged up".  Maybe you should beat around the bush
>>>     less with your "requirements" for your application you are expecting me to
>>>     support for you.
>>>
>>>     I already added 2 patches for you right.  Just be clear about what you want.
>>>
>>>
>>>
>>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     Not necessarily. For instance I got a "send_cancel" when the
>>>>     calling party hanged up before the other party could pick up.
>>>>     Also, shouldn't something like that be protocol/technology
>>>>     neutral?
>>>>
>>>>
>>>>
>>>>     Anthony Minessale wrote:
>>>>
>>>>     sip_hangup_disposition will be set to recv_bye on the side that was
>>>>     hungup.
>>>>
>>>>
>>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     Any updates about the "which side hanged up" potential variable?
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     Makes sense. I will look into this.
>>>>>     -MC
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>     I am sending this second email to request/suggest/enquire about something
>>>>>     relevant :
>>>>>
>>>>>     Wouldn't it be useful to know which end of a specific call leg send the
>>>>>     protocol
>>>>>     specific hangup cause? Otherwise it would be difficult to understand what
>>>>>     really happened.
>>>>>
>>>>>
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     I will do some research on this and let you know what I find out.
>>>>>     Question: are these internal calls or pstn or ?? Just curious about
>>>>>     your environment.
>>>>>
>>>>>     Thanks,
>>>>>     MC
>>>>>
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>
>>>>>
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>     ________________________________
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>         
>>     _______________________________________________
>>     Freeswitch-users mailing list
>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From regs at kinetix.gr  Tue Dec  9 08:39:57 2008
From: regs at kinetix.gr (regs at kinetix.gr)
Date: Tue, 09 Dec 2008 18:39:57 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>
	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
Message-ID: <493E9F5D.7020906@kinetix.gr>

That approach introduces a third party application to
the setup (in order to capture and parse tha SIP messages)
that adds a lot in terms of complexity and reliability ( and cpu
usage). Also it could become a nightmare when you use a
mix of protocols (iax, sip, h323) and technologies (openzap etc).

In the case of a live debugging session, capturing is the most useful tool
but if you want to troubleshoot based on historical data (CDRs) then you
need some detailing. In addition you don't have to fill your databases
with all the fields that FS gives you in an XML cdr. You could
only pick those which are of interest in a particular application.

Shelby Ramsey wrote:
> Hello,
>
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise 
> can be a little trying (a row in your CDR database could be over 100 
> fields long!).  
>
> The best option here is to catch the UUID's for the 2 call legs, 
> capture all SIP messaging, parse and dump the messaging, and then 
> correlate the calls from the CDR from there.  
>
> Much easier than trying to do it from FS ... and most folks want to 
> see SIP captures anyway (very broad set of tools to debug).  
>
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.
>
>
>
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos 
> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>
>
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
>
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be
>     introduced
>     to FS come from real life issues that we have faced during the
>     last few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
>
>
>
>
>     Michael Collins wrote:
>>     Thanks for your feedback. It definitely helps to know not only what
>>     you need FS to do but why you need it to do so.
>>
>>     Do you have FS in production right now? Just curious.
>>
>>     Thanks,
>>     MC
>>
>>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "I already added 2 patches for you right.  Just be clear about what you
>>>     want."
>>>
>>>     And I am grateful of that.
>>>
>>>     "it is protocol neutral, that's why it starts with sip_"
>>>
>>>     I didn't know that. I thought that the sip_ variables are protocol specific.
>>>     So one would expect there to be an iax_hangup_disposition,
>>>     woomera_hangup_disposition etc?
>>>
>>>     "Maybe you should beat around the bush less with your "requirements" for
>>>     your application you are expecting me to support for you."
>>>
>>>     I am just trying to gather statistics for my providers as I would with any
>>>     VoIP softswitch. (hangup causes per terminator per destination)
>>>     I don't think that this is a specific "application" rather than a general
>>>     necessity for VoIP carriers. It is also very useful for troubleshooting
>>>     purposes : when I look at my CDRs to find a call that I got a complain for,
>>>     I want to be able to tell if it was me or the provider who
>>>     hanged up and gave a specific hangup cause, so that I can troubleshoot the
>>>     issue better.
>>>
>>>     "Just be clear about what you want."
>>>
>>>     I want FS to reach that level of detailing and maturity in all aspects so
>>>     that it could be the softswitch of choice by any VoIP entrepreneur
>>>     (or hobbyist) and it is my strong belief that this can only be done by the
>>>     community giving feedback to the programmers about what
>>>     they find useful or not (i.e. experience from real-life situations). The
>>>     patches that you made the last few days were not intended for
>>>     me exclusively but for anyone that will face the same situations using FS.
>>>     If you want the community to stop sending feedback about
>>>     features/improvements you may as well close down this mailing list or just
>>>     use it as an announcement board.
>>>
>>>     I wish I was a c programmer and get involved with the project actively. But
>>>     I am not. And as far as I can tell most of the registered users
>>>     in this list aren't either. So they only way we can help is by testing and
>>>     suggesting.
>>>
>>>     Anthony Minessale wrote:
>>>
>>>     it is protocol neutral, that's why it starts with sip_
>>>
>>>     the variable can be any of:
>>>
>>>     send_bye
>>>     recv_bye
>>>     send_cancel
>>>     send_refuse
>>>
>>>
>>>     using that value you can determine the information you asked.  I answered
>>>     your specific question which was:
>>>     determining "which side hanged up".  Maybe you should beat around the bush
>>>     less with your "requirements" for your application you are expecting me to
>>>     support for you.
>>>
>>>     I already added 2 patches for you right.  Just be clear about what you want.
>>>
>>>
>>>
>>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     Not necessarily. For instance I got a "send_cancel" when the
>>>>     calling party hanged up before the other party could pick up.
>>>>     Also, shouldn't something like that be protocol/technology
>>>>     neutral?
>>>>
>>>>
>>>>
>>>>     Anthony Minessale wrote:
>>>>
>>>>     sip_hangup_disposition will be set to recv_bye on the side that was
>>>>     hungup.
>>>>
>>>>
>>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     Any updates about the "which side hanged up" potential variable?
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     Makes sense. I will look into this.
>>>>>     -MC
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>     I am sending this second email to request/suggest/enquire about something
>>>>>     relevant :
>>>>>
>>>>>     Wouldn't it be useful to know which end of a specific call leg send the
>>>>>     protocol
>>>>>     specific hangup cause? Otherwise it would be difficult to understand what
>>>>>     really happened.
>>>>>
>>>>>
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     I will do some research on this and let you know what I find out.
>>>>>     Question: are these internal calls or pstn or ?? Just curious about
>>>>>     your environment.
>>>>>
>>>>>     Thanks,
>>>>>     MC
>>>>>
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>
>>>>>
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>     ________________________________
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>         
>>     _______________________________________________
>>     Freeswitch-users mailing list
>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
>
>
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From vhatz at kinetix.gr  Tue Dec  9 08:42:54 2008
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Tue, 09 Dec 2008 18:42:54 +0200
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
References: <49391D40.6050103@kinetix.gr>
	<49395403.6080404@kinetix.gr>	<5B9E89AE-1F6D-49B0-9E27-73CD156DD345@freeswitch.org>	<493CF2D4.6010904@kinetix.gr>	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>	<493D2B9B.6050205@kinetix.gr>	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>	<493E2A74.7010502@kinetix.gr>	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
Message-ID: <493EA00E.7070907@kinetix.gr>

Shelby Ramsey wrote:
> Hello,
> 
> This is just my 2 cents ... but my experience has been that trying to 
> catch all of the various variables (i.e. from XML_CDR) or otherwise can 
> be a little trying (a row in your CDR database could be over 100 fields 
> long!).  
> 
> The best option here is to catch the UUID's for the 2 call legs, capture 
> all SIP messaging, parse and dump the messaging, and then correlate the 
> calls from the CDR from there.  
> 
> Much easier than trying to do it from FS ... and most folks want to see 
> SIP captures anyway (very broad set of tools to debug).  
> 
> Measuring things like ASR, PDD, etc in my opinion is much easier from 
> the raw messaging than trying to do something with FS CDR records.

That can certainly be an option, especially for debugging purposes.

However, under heavy load (imagine a few thousands of calls per hour, a 
few millions per day) logging and parsing all the SIP messages on file 
will be a problem.

Also, logging SIP messages is oriented to SIP only, when a more protocol 
agnostic approach could be followed. Plus, we would still need to parse 
a lot of text to extract the information that we need, while in a CDR 
(even a long one with many fields) we have a lot of information with a 
minimum hassle.

We've seen in production environments that excessive logging wastes I/O 
power and disk space, this is why (we at least) turn it on in our 
various systems only when we need it for troubleshooting, and 
immediately turn it off afterwards.

Additionally, a very long CDR is a lot less text to write on disk once, 
after the call is over, rather than writing many, whole packets during 
the duration of a call.

A 100-field-CDR on file could not be much of a problem, because usually 
these the raw CDR fields are rarely imported in a database in their 
entirety for billing or QoS analysis. A lot of the information which is 
not used directly & immediately for billing or QoS analysis remains on 
file in case needs to do basic troubleshooting in arrears. Granted, we 
would not have the same amount of information as with the written SIP 
messages, but it is useful nonetheless.

Of course, I need to stress that I write all this coming from the 
background of VoIP carriers. The above could apply well for typical & 
simple scenarios, where a call leg comes into FS and another calls leg 
comes out of it, which is what most carriers do.

If we need billing and QoS analysis for IVR's, queues, call transfers, 
etc, then yes, one-line CDRs would not do. In this case, logging whole 
packets could be a solution, although an event-based approach could be 
much better to cover all protocols/technologies (IAX, TDM cards, etc), IMHO.

Best regards,
Vlasis Hatzistavrou
Kinetix Tele.com Hellas Ltd.
Monastiriou 9 & Enotikon
54627
Thessaloniki
Greece
Tel.: +302310556134
Fax: +302310556134 (ext. 0)
GSM: +306977835653
e-mail: vhatz at kinetix.gr
http://www.kinetix.gr



> 
> 
> 
> On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs at kinetix.gr 
> <mailto:regs at kinetix.gr>> wrote:
> 
> 
>         We are currently in the migration process from our
>     current system to a FS based setup. We are in the process of
>     adapting our billing and routing to FS. All the  CDRs (and variables)
>     related issues that we have been discussing on this mailing list
>     come from the need to extract the same level of information from FS as
>     we do with our current closed source proprietary system. So, we
>     chose FS because of the versatility it provides in every aspect (event
>     handling, config implementation etc.) and we strongly believe that all
>     these additions/fixes would be beneficial to many potential FS users.
> 
>         We are at your disposal for more details in case you need
>     more information about what exactly we are trying to do. Basically,
>     our approach is from the VoIP carrier's point of view rather than the
>     PBX user's/implementor's. So, the details that we asked to be introduced
>     to FS come from real life issues that we have faced during the last
>     few years
>     with various platforms and troubleshooting experiences with other
>     VoIP carriers.
> 
> 
> 
> 
>     Michael Collins wrote:
>>     Thanks for your feedback. It definitely helps to know not only what
>>     you need FS to do but why you need it to do so.
>>
>>     Do you have FS in production right now? Just curious.
>>
>>     Thanks,
>>     MC
>>
>>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
>>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
>>       
>>>     "I already added 2 patches for you right.  Just be clear about what you
>>>     want."
>>>
>>>     And I am grateful of that.
>>>
>>>     "it is protocol neutral, that's why it starts with sip_"
>>>
>>>     I didn't know that. I thought that the sip_ variables are protocol specific.
>>>     So one would expect there to be an iax_hangup_disposition,
>>>     woomera_hangup_disposition etc?
>>>
>>>     "Maybe you should beat around the bush less with your "requirements" for
>>>     your application you are expecting me to support for you."
>>>
>>>     I am just trying to gather statistics for my providers as I would with any
>>>     VoIP softswitch. (hangup causes per terminator per destination)
>>>     I don't think that this is a specific "application" rather than a general
>>>     necessity for VoIP carriers. It is also very useful for troubleshooting
>>>     purposes : when I look at my CDRs to find a call that I got a complain for,
>>>     I want to be able to tell if it was me or the provider who
>>>     hanged up and gave a specific hangup cause, so that I can troubleshoot the
>>>     issue better.
>>>
>>>     "Just be clear about what you want."
>>>
>>>     I want FS to reach that level of detailing and maturity in all aspects so
>>>     that it could be the softswitch of choice by any VoIP entrepreneur
>>>     (or hobbyist) and it is my strong belief that this can only be done by the
>>>     community giving feedback to the programmers about what
>>>     they find useful or not (i.e. experience from real-life situations). The
>>>     patches that you made the last few days were not intended for
>>>     me exclusively but for anyone that will face the same situations using FS.
>>>     If you want the community to stop sending feedback about
>>>     features/improvements you may as well close down this mailing list or just
>>>     use it as an announcement board.
>>>
>>>     I wish I was a c programmer and get involved with the project actively. But
>>>     I am not. And as far as I can tell most of the registered users
>>>     in this list aren't either. So they only way we can help is by testing and
>>>     suggesting.
>>>
>>>     Anthony Minessale wrote:
>>>
>>>     it is protocol neutral, that's why it starts with sip_
>>>
>>>     the variable can be any of:
>>>
>>>     send_bye
>>>     recv_bye
>>>     send_cancel
>>>     send_refuse
>>>
>>>
>>>     using that value you can determine the information you asked.  I answered
>>>     your specific question which was:
>>>     determining "which side hanged up".  Maybe you should beat around the bush
>>>     less with your "requirements" for your application you are expecting me to
>>>     support for you.
>>>
>>>     I already added 2 patches for you right.  Just be clear about what you want.
>>>
>>>
>>>
>>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>     wrote:
>>>         
>>>>     Not necessarily. For instance I got a "send_cancel" when the
>>>>     calling party hanged up before the other party could pick up.
>>>>     Also, shouldn't something like that be protocol/technology
>>>>     neutral?
>>>>
>>>>
>>>>
>>>>     Anthony Minessale wrote:
>>>>
>>>>     sip_hangup_disposition will be set to recv_bye on the side that was
>>>>     hungup.
>>>>
>>>>
>>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>     wrote:
>>>>           
>>>>>     Any updates about the "which side hanged up" potential variable?
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     Makes sense. I will look into this.
>>>>>     -MC
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>     I am sending this second email to request/suggest/enquire about something
>>>>>     relevant :
>>>>>
>>>>>     Wouldn't it be useful to know which end of a specific call leg send the
>>>>>     protocol
>>>>>     specific hangup cause? Otherwise it would be difficult to understand what
>>>>>     really happened.
>>>>>
>>>>>
>>>>>
>>>>>     Michael S Collins wrote:
>>>>>
>>>>>     I will do some research on this and let you know what I find out.
>>>>>     Question: are these internal calls or pstn or ?? Just curious about
>>>>>     your environment.
>>>>>
>>>>>     Thanks,
>>>>>     MC
>>>>>
>>>>>
>>>>>
>>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <regs at kinetix.gr> <mailto:regs at kinetix.gr>
>>>>>     wrote:
>>>>>
>>>>>
>>>>>
>>>>>     The proto_specific_hangup_cause is missing on one of the two
>>>>>     call legs. When the caller hangs up it is missing from the a-leg CDR.
>>>>>     When the callee hangs up it is missing from the b-leg CDR. Is this
>>>>>     nornal?
>>>>>
>>>>>     And another question : what piece of info could inform me about who
>>>>>     hanged up?
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>     ________________________________
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>
>>>>>     --
>>>>>     -------------------------------------------
>>>>>     Apostolos Pantsiopoulos
>>>>>     Kinetix Tele.com R & D
>>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>>     -------------------------------------------
>>>>>
>>>>>     _______________________________________________
>>>>>     Freeswitch-users mailing list
>>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org
>>>>>
>>>>>             
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>>     pstn:213-799-1400
>>>>
>>>>     ________________________________
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>
>>>>     --
>>>>     -------------------------------------------
>>>>     Apostolos Pantsiopoulos
>>>>     Kinetix Tele.com R & D
>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>>     -------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     Freeswitch-users mailing list
>>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>>           
>>>     --
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com <mailto:MSN:anthony_minessale at hotmail.com>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com <mailto:GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>
>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:guest at conference.freeswitch.org/888>
>>>     googletalk:conf+888 at conference.freeswitch.org <mailto:googletalk:conf+888 at conference.freeswitch.org>
>>>     pstn:213-799-1400
>>>
>>>     ________________________________
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>     --
>>>     -------------------------------------------
>>>     Apostolos Pantsiopoulos
>>>     Kinetix Tele.com R & D
>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>>>     -------------------------------------------
>>>
>>>     _______________________________________________
>>>     Freeswitch-users mailing list
>>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>>         
>>     _______________________________________________
>>     Freeswitch-users mailing list
>>     Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>       
> 
> 
>     -- 
>     -------------------------------------------
>     Apostolos Pantsiopoulos
>     Kinetix Tele.com R & D
>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
>     ------------------------------------------- 
> 
> 
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From woof at nortel.com  Tue Dec  9 08:45:03 2008
From: woof at nortel.com (Andy Spitzer)
Date: Tue, 09 Dec 2008 11:45:03 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
Message-ID: <op.ulwl5dbessieo1@woof.us.nortel.com>

Woof!

It appears that FreeSWITCH writes 

   freeswitch.history
   freeswitch.log
   freeswitch.pid
   freeswitch.xml.fsxml

to the -log directory.  

Is there a way to put the files other than freeswitch.log into the -db directory instead?

In my environment we archive and rotate everything in the log directory (which includes logs beside FreeSWITCH's), and these other FreeSWITCH files are getting rotated.  Yeah, I can explicitly exclude them, but to me it seems those really belong in the -db directory anyway, as they are inherently data needed for the current executable of FreeSWITCH, and not logs.

--Woof!



From mrjoebain at gmail.com  Tue Dec  9 08:53:30 2008
From: mrjoebain at gmail.com (Joe Bain)
Date: Tue, 9 Dec 2008 16:53:30 +0000
Subject: [Freeswitch-users] Catching hangups
In-Reply-To: <8F96510A-1A1E-45E9-A206-FDA66CAEA06F@jerris.com>
References: <748d53500812080857u51b9f963v3b7ff28d334d6660@mail.gmail.com>
	<87f2f3b90812080913q2d9d721dm414983c781fade3d@mail.gmail.com>
	<748d53500812090127x32d924a9m1d6c009a235bbf4b@mail.gmail.com>
	<1FDD74F3-5BBE-45B6-954D-EE8C21D815FA@myrvold.org>
	<748d53500812090206l7a65b2a4w400c34b87c9e22e7@mail.gmail.com>
	<748d53500812090610t70a11a07u594541a8e132a9d3@mail.gmail.com>
	<8F96510A-1A1E-45E9-A206-FDA66CAEA06F@jerris.com>
Message-ID: <748d53500812090853v2339ac3x2ca77446884eab9@mail.gmail.com>

2008/12/9 Michael Jerris <mike at jerris.com>

>
>  On Dec 9, 2008, at 9:10 AM, Joe Bain wrote:
>
>  Ok I have been testing more and I have reduced my problem to a pretty
> short and simple Lua script. I've posted it at
> http://pastebin.freeswitch.org/6373 and this gets called straight from the
> dialplan. From my experience so far it only exits after a caller hangup
> about 1 in 10 times. Most of the time it continues to loop until I do
> 'hupall'.
>
> Thanks in advance if anyone can solve this or offer any advice.
>
> Joe
>
> 2008/12/9 Joe Bain <mrjoebain at gmail.com>
>
>> 2008/12/9 Ivan C Myrvold <ivan at myrvold.org>
>>
>>  Did you read carefully when asked to provide login and password? The
>>> login and password is there, don't use your own freeswitch login.
>>>
>>> Ivan
>>>
>>>  Den 9. des.. 2008 kl. 10:27 skrev Joe Bain:
>>>
>>>  On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <mrjoebain at gmail.com> wrote:
>>> > Hi,
>>> >
>>> > I'm writing an IVR in Lua and am having problems dealing with hangups
>>> > cleanly. Very often session:ready() reports true long after I have hung
>>> up
>>> > and the hangup hook function I have set doesn't get called either. It
>>> seems
>>> > to report that the session is active indefinitely in some cases where a
>>> loop
>>> > keeps trying to get some dtmf key presses. Is there any trick to using
>>> > session:ready() or the hangup hook that I might have missed?
>>> >
>>> > On a slightly related point I can't seem to access the session
>>> properties,
>>> > e.g. session.caller_id_num has a value of nil. Any thoughts here?
>>> >
>>>
>>>
>        Joe,
>>>
>>>>
>>>> A few questions... what svn rev are you running? Which operating
>>>> system? Finally, is it possible for you to put your dialplan and Lua
>>>> script up at pastebin.freeswitch.org?
>>>>
>>>> Thanks,
>>>> MC
>>>>
>>> Hi,
>>>
>>> I'm running "FreeSWITCH Version 1.0.trunk (10175M)" on Windows. I can
>>> post the dialplan and lua script though at the moment I can't seem to log in
>>> to the pastebin, I just became a member on the freeswitch homepage but the
>>> pass/username isn't being accepted.
>>>
>>> Joe
>>>
>>>
>> Ah, I should have read more carefully!
>>
>> The dialplan is here <http://pastebin.freeswitch.org/6372> and the two
>> important lua scripts are here <http://pastebin.freeswitch.org/6371> and
>> here <http://pastebin.freeswitch.org/6370>, the first calls the second. I
>> didn't include all the Lua script as the problem appears right at the start
>> (as well as throughout) if the user hangs up when the IVR is speaking
>> (asking for an id number) then it seems to never get a hangup and loops
>> trying to get the id number.
>>
>> Joe
>>
>
> We just tested this with current svn trunk and it appears to work fine,
> could you try updating and see if it is still a problem for you
>
> Mike
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
I have to install FS on our server soon so when I do I'll see if the problem
is still there. Though I'm testing on Vista and the server won't be running
that so if the problem doesn't reappear it may not be conclusive. If I have
any spare time I'll try a reinstall on my test machine but I probably won't
unfortunately, my contract ends on wednesday at the company I'm working for.

Thanks for your help.

Joe
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From anthony.minessale at gmail.com  Tue Dec  9 09:08:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Dec 2008 11:08:51 -0600
Subject: [Freeswitch-users] Proto specific hangup cause issue
In-Reply-To: <493EA00E.7070907@kinetix.gr>
References: <49391D40.6050103@kinetix.gr> <493CF2D4.6010904@kinetix.gr>
	<191c3a030812080548ud08e863o8d3990535a8d0f22@mail.gmail.com>
	<493D2B9B.6050205@kinetix.gr>
	<191c3a030812080752k2093fa5dq68ec8d312ccee69e@mail.gmail.com>
	<493E2A74.7010502@kinetix.gr>
	<87f2f3b90812090644i304d72e7u61dc397ab897df50@mail.gmail.com>
	<493E8C7E.1050306@kinetix.gr>
	<35b355e90812090812o2588bfe2gf2daf6c01b2fbc00@mail.gmail.com>
	<493EA00E.7070907@kinetix.gr>
Message-ID: <191c3a030812090908kca14327v821d78998e8b50b6@mail.gmail.com>

see this is better.

That's why I asked you to be more specific about what you want because the
tiny back and
forth questions were not exposing your intent or needs at all.  I answer
every email I can and when threads start to grow without getting to the
point i start to get frustrated.

Now that you have opened up the discussion you have more people chiming in
on the topic.

Yes the sip_* variables are unique to SIP and the one that says
proto_specific are all
done per implementation.

If you would like to suggest a list of standard variables you think apply to
all calls
or something you feel is missing, we can consider injecting them into the
code.




On Tue, Dec 9, 2008 at 10:42 AM, Vlasis Hatzistavrou (KTI) <vhatz at kinetix.gr
> wrote:

> Shelby Ramsey wrote:
> > Hello,
> >
> > This is just my 2 cents ... but my experience has been that trying to
> > catch all of the various variables (i.e. from XML_CDR) or otherwise can
> > be a little trying (a row in your CDR database could be over 100 fields
> > long!).
> >
> > The best option here is to catch the UUID's for the 2 call legs, capture
> > all SIP messaging, parse and dump the messaging, and then correlate the
> > calls from the CDR from there.
> >
> > Much easier than trying to do it from FS ... and most folks want to see
> > SIP captures anyway (very broad set of tools to debug).
> >
> > Measuring things like ASR, PDD, etc in my opinion is much easier from
> > the raw messaging than trying to do something with FS CDR records.
>
> That can certainly be an option, especially for debugging purposes.
>
> However, under heavy load (imagine a few thousands of calls per hour, a
> few millions per day) logging and parsing all the SIP messages on file
> will be a problem.
>
> Also, logging SIP messages is oriented to SIP only, when a more protocol
> agnostic approach could be followed. Plus, we would still need to parse
> a lot of text to extract the information that we need, while in a CDR
> (even a long one with many fields) we have a lot of information with a
> minimum hassle.
>
> We've seen in production environments that excessive logging wastes I/O
> power and disk space, this is why (we at least) turn it on in our
> various systems only when we need it for troubleshooting, and
> immediately turn it off afterwards.
>
> Additionally, a very long CDR is a lot less text to write on disk once,
> after the call is over, rather than writing many, whole packets during
> the duration of a call.
>
> A 100-field-CDR on file could not be much of a problem, because usually
> these the raw CDR fields are rarely imported in a database in their
> entirety for billing or QoS analysis. A lot of the information which is
> not used directly & immediately for billing or QoS analysis remains on
> file in case needs to do basic troubleshooting in arrears. Granted, we
> would not have the same amount of information as with the written SIP
> messages, but it is useful nonetheless.
>
> Of course, I need to stress that I write all this coming from the
> background of VoIP carriers. The above could apply well for typical &
> simple scenarios, where a call leg comes into FS and another calls leg
> comes out of it, which is what most carriers do.
>
> If we need billing and QoS analysis for IVR's, queues, call transfers,
> etc, then yes, one-line CDRs would not do. In this case, logging whole
> packets could be a solution, although an event-based approach could be
> much better to cover all protocols/technologies (IAX, TDM cards, etc),
> IMHO.
>
> Best regards,
> Vlasis Hatzistavrou
> Kinetix Tele.com Hellas Ltd.
> Monastiriou 9 & Enotikon
> 54627
> Thessaloniki
> Greece
> Tel.: +302310556134
> Fax: +302310556134 (ext. 0)
> GSM: +306977835653
> e-mail: vhatz at kinetix.gr
> http://www.kinetix.gr
>
>
>
> >
> >
> >
> > On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos <regs at kinetix.gr
> > <mailto:regs at kinetix.gr>> wrote:
> >
> >
> >         We are currently in the migration process from our
> >     current system to a FS based setup. We are in the process of
> >     adapting our billing and routing to FS. All the  CDRs (and variables)
> >     related issues that we have been discussing on this mailing list
> >     come from the need to extract the same level of information from FS
> as
> >     we do with our current closed source proprietary system. So, we
> >     chose FS because of the versatility it provides in every aspect
> (event
> >     handling, config implementation etc.) and we strongly believe that
> all
> >     these additions/fixes would be beneficial to many potential FS users.
> >
> >         We are at your disposal for more details in case you need
> >     more information about what exactly we are trying to do. Basically,
> >     our approach is from the VoIP carrier's point of view rather than the
> >     PBX user's/implementor's. So, the details that we asked to be
> introduced
> >     to FS come from real life issues that we have faced during the last
> >     few years
> >     with various platforms and troubleshooting experiences with other
> >     VoIP carriers.
> >
> >
> >
> >
> >     Michael Collins wrote:
> >>     Thanks for your feedback. It definitely helps to know not only what
> >>     you need FS to do but why you need it to do so.
> >>
> >>     Do you have FS in production right now? Just curious.
> >>
> >>     Thanks,
> >>     MC
> >>
> >>     On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
> >>     <regs at kinetix.gr> <mailto:regs at kinetix.gr> wrote:
> >>
> >>>     "I already added 2 patches for you right.  Just be clear about what
> you
> >>>     want."
> >>>
> >>>     And I am grateful of that.
> >>>
> >>>     "it is protocol neutral, that's why it starts with sip_"
> >>>
> >>>     I didn't know that. I thought that the sip_ variables are protocol
> specific.
> >>>     So one would expect there to be an iax_hangup_disposition,
> >>>     woomera_hangup_disposition etc?
> >>>
> >>>     "Maybe you should beat around the bush less with your
> "requirements" for
> >>>     your application you are expecting me to support for you."
> >>>
> >>>     I am just trying to gather statistics for my providers as I would
> with any
> >>>     VoIP softswitch. (hangup causes per terminator per destination)
> >>>     I don't think that this is a specific "application" rather than a
> general
> >>>     necessity for VoIP carriers. It is also very useful for
> troubleshooting
> >>>     purposes : when I look at my CDRs to find a call that I got a
> complain for,
> >>>     I want to be able to tell if it was me or the provider who
> >>>     hanged up and gave a specific hangup cause, so that I can
> troubleshoot the
> >>>     issue better.
> >>>
> >>>     "Just be clear about what you want."
> >>>
> >>>     I want FS to reach that level of detailing and maturity in all
> aspects so
> >>>     that it could be the softswitch of choice by any VoIP entrepreneur
> >>>     (or hobbyist) and it is my strong belief that this can only be done
> by the
> >>>     community giving feedback to the programmers about what
> >>>     they find useful or not (i.e. experience from real-life
> situations). The
> >>>     patches that you made the last few days were not intended for
> >>>     me exclusively but for anyone that will face the same situations
> using FS.
> >>>     If you want the community to stop sending feedback about
> >>>     features/improvements you may as well close down this mailing list
> or just
> >>>     use it as an announcement board.
> >>>
> >>>     I wish I was a c programmer and get involved with the project
> actively. But
> >>>     I am not. And as far as I can tell most of the registered users
> >>>     in this list aren't either. So they only way we can help is by
> testing and
> >>>     suggesting.
> >>>
> >>>     Anthony Minessale wrote:
> >>>
> >>>     it is protocol neutral, that's why it starts with sip_
> >>>
> >>>     the variable can be any of:
> >>>
> >>>     send_bye
> >>>     recv_bye
> >>>     send_cancel
> >>>     send_refuse
> >>>
> >>>
> >>>     using that value you can determine the information you asked.  I
> answered
> >>>     your specific question which was:
> >>>     determining "which side hanged up".  Maybe you should beat around
> the bush
> >>>     less with your "requirements" for your application you are
> expecting me to
> >>>     support for you.
> >>>
> >>>     I already added 2 patches for you right.  Just be clear about what
> you want.
> >>>
> >>>
> >>>
> >>>     On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos <
> regs at kinetix.gr> <mailto:regs at kinetix.gr>
> >>>     wrote:
> >>>
> >>>>     Not necessarily. For instance I got a "send_cancel" when the
> >>>>     calling party hanged up before the other party could pick up.
> >>>>     Also, shouldn't something like that be protocol/technology
> >>>>     neutral?
> >>>>
> >>>>
> >>>>
> >>>>     Anthony Minessale wrote:
> >>>>
> >>>>     sip_hangup_disposition will be set to recv_bye on the side that
> was
> >>>>     hungup.
> >>>>
> >>>>
> >>>>     On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos <
> regs at kinetix.gr> <mailto:regs at kinetix.gr>
> >>>>     wrote:
> >>>>
> >>>>>     Any updates about the "which side hanged up" potential variable?
> >>>>>
> >>>>>     Michael S Collins wrote:
> >>>>>
> >>>>>     Makes sense. I will look into this.
> >>>>>     -MC
> >>>>>
> >>>>>
> >>>>>     On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos <
> regs at kinetix.gr> <mailto:regs at kinetix.gr>
> >>>>>     wrote:
> >>>>>
> >>>>>     I am sending this second email to request/suggest/enquire about
> something
> >>>>>     relevant :
> >>>>>
> >>>>>     Wouldn't it be useful to know which end of a specific call leg
> send the
> >>>>>     protocol
> >>>>>     specific hangup cause? Otherwise it would be difficult to
> understand what
> >>>>>     really happened.
> >>>>>
> >>>>>
> >>>>>
> >>>>>     Michael S Collins wrote:
> >>>>>
> >>>>>     I will do some research on this and let you know what I find out.
> >>>>>     Question: are these internal calls or pstn or ?? Just curious
> about
> >>>>>     your environment.
> >>>>>
> >>>>>     Thanks,
> >>>>>     MC
> >>>>>
> >>>>>
> >>>>>
> >>>>>     On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos <
> regs at kinetix.gr> <mailto:regs at kinetix.gr>
> >>>>>     wrote:
> >>>>>
> >>>>>
> >>>>>
> >>>>>     The proto_specific_hangup_cause is missing on one of the two
> >>>>>     call legs. When the caller hangs up it is missing from the a-leg
> CDR.
> >>>>>     When the callee hangs up it is missing from the b-leg CDR. Is
> this
> >>>>>     nornal?
> >>>>>
> >>>>>     And another question : what piece of info could inform me about
> who
> >>>>>     hanged up?
> >>>>>
> >>>>>
> >>>>>     --
> >>>>>     -------------------------------------------
> >>>>>     Apostolos Pantsiopoulos
> >>>>>     Kinetix Tele.com R & D
> >>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>>>     -------------------------------------------
> >>>>>
> >>>>>
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>     --
> >>>>>     -------------------------------------------
> >>>>>     Apostolos Pantsiopoulos
> >>>>>     Kinetix Tele.com R & D
> >>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>>>     -------------------------------------------
> >>>>>
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>     ________________________________
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>     --
> >>>>>     -------------------------------------------
> >>>>>     Apostolos Pantsiopoulos
> >>>>>     Kinetix Tele.com R & D
> >>>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>>>     -------------------------------------------
> >>>>>
> >>>>>     _______________________________________________
> >>>>>     Freeswitch-users mailing list
> >>>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>     http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>     --
> >>>>     Anthony Minessale II
> >>>>
> >>>>     FreeSWITCH http://www.freeswitch.org/
> >>>>     ClueCon http://www.cluecon.com/
> >>>>
> >>>>     AIM: anthm
> >>>>     MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com><mailto:
> MSN:anthony_minessale at hotmail.com>
> >>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com><mailto:
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> >>>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>>>
> >>>>     FreeSWITCH Developer Conference
> >>>>     sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:
> guest at conference.freeswitch.org/888>
> >>>>     googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org><mailto:
> googletalk:conf+888 at conference.freeswitch.org<conf%2B888 at conference.freeswitch.org>
> >
> >>>>     pstn:213-799-1400
> >>>>
> >>>>     ________________________________
> >>>>     _______________________________________________
> >>>>     Freeswitch-users mailing list
> >>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>     http://www.freeswitch.org
> >>>>
> >>>>
> >>>>     --
> >>>>     -------------------------------------------
> >>>>     Apostolos Pantsiopoulos
> >>>>     Kinetix Tele.com R & D
> >>>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>>     -------------------------------------------
> >>>>
> >>>>     _______________________________________________
> >>>>     Freeswitch-users mailing list
> >>>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>     http://www.freeswitch.org
> >>>>
> >>>>
> >>>     --
> >>>     Anthony Minessale II
> >>>
> >>>     FreeSWITCH http://www.freeswitch.org/
> >>>     ClueCon http://www.cluecon.com/
> >>>
> >>>     AIM: anthm
> >>>     MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com><mailto:
> MSN:anthony_minessale at hotmail.com>
> >>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com><mailto:
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >
> >>>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>>
> >>>     FreeSWITCH Developer Conference
> >>>     sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>>     iax:guest at conference.freeswitch.org/888 <mailto:iax:
> guest at conference.freeswitch.org/888>
> >>>     googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org><mailto:
> googletalk:conf+888 at conference.freeswitch.org<conf%2B888 at conference.freeswitch.org>
> >
> >>>     pstn:213-799-1400
> >>>
> >>>     ________________________________
> >>>     _______________________________________________
> >>>     Freeswitch-users mailing list
> >>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>     http://www.freeswitch.org
> >>>
> >>>
> >>>     --
> >>>     -------------------------------------------
> >>>     Apostolos Pantsiopoulos
> >>>     Kinetix Tele.com R & D
> >>>     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >>>     -------------------------------------------
> >>>
> >>>     _______________________________________________
> >>>     Freeswitch-users mailing list
> >>>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>     http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>     _______________________________________________
> >>     Freeswitch-users mailing list
> >>     Freeswitch-users at lists.freeswitch.org <mailto:
> Freeswitch-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     http://www.freeswitch.org
> >>
> >
> >
> >     --
> >     -------------------------------------------
> >     Apostolos Pantsiopoulos
> >     Kinetix Tele.com R & D
> >     email: regs at kinetix.gr <mailto:regs at kinetix.gr>
> >     -------------------------------------------
> >
> >
> >     _______________________________________________
> >     Freeswitch-users mailing list
> >     Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From erick at junctionnetworks.com  Tue Dec  9 10:06:38 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 13:06:38 -0500
Subject: [Freeswitch-users]  Sending SIP calls via outbound proxy
References: 493DC658.8020305@junctionnetworks.com
Message-ID: <493EB3AE.1090109@junctionnetworks.com>

Hi Brian,

Thanks for the reply, but I still don't think that answers my original
question.  I'm trying to get FS to act simply as a UAC in this
instance, what I want is for FS to proxy ALL outbound calls through my
proxy server at foo.com.

So when FS originates a call to alice at bar.com I want the signaling path
to be set up as:

FreeSwitch ---> proxy.foo.com ---> alice at bar.com

I found this thread: 
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/008582.html
but I still can't seem to get FS to stop resolving domain bar.com on 
it's own, even when I set the
sip_invite_domain variable like so:
 originate 
{sip_invite_domain='proxy.foo.com'}sofia/external/alice at bar.com &echo()

That is how I ended up using the "originate 
sofia/gateway/proxy/alice at bar.com &echo()" command.
While I understand that's wrong, I don't know what the right config/cmd 
is to accomplish my task.

Any other help is much appreciated.

Thanks,

Erick


>
> First example is WRONG you don't dial via a gateway that way.  If you  
> wish to dial alice at bar.com then try sofia/internal/alice at bar.com 
> as  
> you don't require a gateway to call alice right?
>
> /b
>
> On Dec 8, 2008, at 7:14 PM, Erick Johnson wrote:
>
> > Here is the command that I'm trying to use that behaves unexpectedly:
> > originate sofia/gateway/proxy/alice at bar.com &echo()
> >
> > However this command produces the results I'm expecting:
> > originate sofia/gateway/proxy/15551234 &echo()
>

-- 
Erick Johnson

sip/email erick at junctionnetworks.com
1-800-801-3381 x7006
Software Engineer
Junction Networks




From brian at freeswitch.org  Tue Dec  9 10:15:47 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 12:15:47 -0600
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493EB3AE.1090109@junctionnetworks.com>
References: 493DC658.8020305@junctionnetworks.com
	<493EB3AE.1090109@junctionnetworks.com>
Message-ID: <E5B2EB7B-A125-4861-A23D-2051242E0C1D@freeswitch.org>

Can you clarify why you need a gateway?  Is the far side going to  
challenge us and request authentication credentials?

So you want us to not resolve the domain of the target at all in any  
way?  That kinda breaks the rules because you should always check the  
NAPTR's and SRV and resolve to the target in that manner its a  
requirement.  If you want to force things to a proxy and let the proxy  
on the far side do the work then you do this:

sofia/profile/alice at bar.com;fs_path=proxy.foo.com

/b



On Dec 9, 2008, at 12:06 PM, Erick Johnson wrote:

> Hi Brian,
>
> Thanks for the reply, but I still don't think that answers my original
> question.  I'm trying to get FS to act simply as a UAC in this
> instance, what I want is for FS to proxy ALL outbound calls through my
> proxy server at foo.com.
>
> So when FS originates a call to alice at bar.com I want the signaling  
> path
> to be set up as:
>
> FreeSwitch ---> proxy.foo.com ---> alice at bar.com
>
> I found this thread:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/008582.html
> but I still can't seem to get FS to stop resolving domain bar.com on
> it's own, even when I set the
> sip_invite_domain variable like so:
> originate
> {sip_invite_domain='proxy.foo.com'}sofia/external/alice at bar.com  
> &echo()
>
> That is how I ended up using the "originate
> sofia/gateway/proxy/alice at bar.com &echo()" command.
> While I understand that's wrong, I don't know what the right config/ 
> cmd
> is to accomplish my task.
>
> Any other help is much appreciated.
>
> Thanks,
>
> Erick

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From erick at junctionnetworks.com  Tue Dec  9 11:11:26 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 14:11:26 -0500
Subject: [Freeswitch-users]  Sending SIP calls via outbound proxy
References: 493EB3AE.1090109@junctionnetworks.com
Message-ID: <493EC2DE.2090609@junctionnetworks.com>

Thanks Brian,

I never did want to use a gateway - I was just lost on how to force FS 
to use a proxy.
fs_path seems to be what I'm looking for.  However now what I run my
originate command the channel gets terminated before FS even sends out
a packet.  The api call completes with NORMAL_UNSPECIFIED
termination cause.

API CALL 
[originate(sofia/external/erick at ejjohnson.org;fs_path=proxy.foo.net 
&echo())] output:
-ERR NORMAL_UNSPECIFIED

Looking at the logs the reason as to why it's been termintated isn't cleear
to me.  Any thoughts?

Here is the pastebin for the log http://pastebin.freeswitch.org/6378

Thanks again

E






> Can you clarify why you need a gateway?  Is the far side going to  
> challenge us and request authentication credentials?
>
> So you want us to not resolve the domain of the target at all in any  
> way?  That kinda breaks the rules because you should always check the  
> NAPTR's and SRV and resolve to the target in that manner its a  
> requirement.  If you want to force things to a proxy and let the proxy  
> on the far side do the work then you do this:
>
> sofia/profile/alice at bar.com;fs_path=proxy.foo.com
>
> /b
>
>



From brian at freeswitch.org  Tue Dec  9 11:17:41 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 13:17:41 -0600
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493EC2DE.2090609@junctionnetworks.com>
References: 493EB3AE.1090109@junctionnetworks.com
	<493EC2DE.2090609@junctionnetworks.com>
Message-ID: <71B232D7-BC0C-4028-B768-56960285C85E@freeswitch.org>

I think you need to '' the sofia uri

/b

On Dec 9, 2008, at 1:11 PM, Erick Johnson wrote:

> Looking at the logs the reason as to why it's been termintated isn't  
> cleear
> to me.  Any thoughts?




From erick at junctionnetworks.com  Tue Dec  9 11:41:55 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 14:41:55 -0500
Subject: [Freeswitch-users]  Sending SIP calls via outbound proxy
References: 493EC2DE.2090609@junctionnetworks.com
Message-ID: <493ECA03.2030106@junctionnetworks.com>

Both:

originate sofia/external/'erick at ejjohnson.org;fs_path=proxybeta.foo.net' 
&echo()
originate sofia/external/erick at ejjohnson.org;fs_path=proxybeta.foo.net 
&echo()

produce the exact same result & log

:(

> * I think you need to '' the sofia uri /b 




From msc at freeswitch.org  Tue Dec  9 11:52:59 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 11:52:59 -0800
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493ECA03.2030106@junctionnetworks.com>
References: <493ECA03.2030106@junctionnetworks.com>
Message-ID: <87f2f3b90812091152j43d9659agadf3eba00554c8b2@mail.gmail.com>

What SVN rev are you running? Also, could you do a SIP trace?
TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
Pastebin the output of that and we'll take it from there.
-MC

On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
<erick at junctionnetworks.com> wrote:
> Both:
>
> originate sofia/external/'erick at ejjohnson.org;fs_path=proxybeta.foo.net'
> &echo()
> originate sofia/external/erick at ejjohnson.org;fs_path=proxybeta.foo.net
> &echo()
>
> produce the exact same result & log
>
> :(
>
>> * I think you need to '' the sofia uri /b
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Dec  9 11:56:27 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2008 13:56:27 -0600
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <87f2f3b90812091152j43d9659agadf3eba00554c8b2@mail.gmail.com>
References: <493ECA03.2030106@junctionnetworks.com>
	<87f2f3b90812091152j43d9659agadf3eba00554c8b2@mail.gmail.com>
Message-ID: <06E304A9-C9D0-45AD-A24C-1D438CEB93C2@freeswitch.org>

originate 'sofia/internal/brian at bkw.org;fs_path=bob.com' &echo()

/b

On Dec 9, 2008, at 1:52 PM, Michael Collins wrote:

> What SVN rev are you running? Also, could you do a SIP trace?
> TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
> Pastebin the output of that and we'll take it from there.
> -MC
>
> On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
> <erick at junctionnetworks.com> wrote:
>> Both:
>>
>> originate sofia/ 
>> external/'erick at ejjohnson.org;fs_path=proxybeta.foo.net'
>> &echo()
>> originate sofia/external/ 
>> erick at ejjohnson.org;fs_path=proxybeta.foo.net
>> &echo()
>>
>> produce the exact same result & log
>>
>> :(
>>
>>> * I think you need to '' the sofia uri /b
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From erick at junctionnetworks.com  Tue Dec  9 12:47:08 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 15:47:08 -0500
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
References: 493ECA03.2030106@junctionnetworks.com
Message-ID: <493ED94C.9080508@junctionnetworks.com>

I'm running latest trunk - Revision: 10682

I've been doing an ngrep on my external freeswitch SIP port and FS
is not sending any SIP packets anywhere when I run the following command.
Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are logged.

originate 
'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' &echo()

Also, just to be clear, when I remove ";fs_path=..." from the command 
above a call
is set up normally to erick at ejjohnson.org and the SIP packets are logged 
to console.

Thanks guys.

> What SVN rev are you running? Also, could you do a SIP trace?
> TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
> Pastebin the output of that and we'll take it from there.
> -MC
>
> On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
> <erick at junctionnetworks.com> wrote:
> > Both:
> >
> > originate sofia/external/'erick at 
> ejjohnson.org;fs_path=proxybeta.foo.net'
> > &echo()
> > originate sofia/external/erick at 
> ejjohnson.org;fs_path=proxybeta.foo.net
> > &echo()
> >
> > produce the exact same result & log
> >
> > :(
> >
> >> * I think you need to '' the sofia uri /b
> >
> >




From erick at junctionnetworks.com  Tue Dec  9 12:49:42 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Tue, 09 Dec 2008 15:49:42 -0500
Subject: [Freeswitch-users]  Sending SIP calls via outbound proxy
Message-ID: <493ED9E6.3000803@junctionnetworks.com>


i forgot to give you the pastebin URL
http://pastebin.freeswitch.org/6379

>
> I'm running latest trunk - Revision: 10682
>
> I've been doing an ngrep on my external freeswitch SIP port and FS
> is not sending any SIP packets anywhere when I run the following command.
> Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are 
> logged.
>
> originate
> 'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' 
> &echo()
>
> Also, just to be clear, when I remove ";fs_path=..." from the command
> above a call
> is set up normally to erick at ejjohnson.org and the SIP packets are 
> logged
> to console.
>
> Thanks guys.
>
> > What SVN rev are you running? Also, could you do a SIP trace?
> > TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
> > Pastebin the output of that and we'll take it from there.
> > -MC
> >
> > On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
> > <erick at junctionnetworks.com> wrote:
> > > Both:
> > >
> > > originate sofia/external/'erick at
> > ejjohnson.org;fs_path=proxybeta.foo.net'
> > > &echo()
> > > originate sofia/external/erick at
> > ejjohnson.org;fs_path=proxybeta.foo.net
> > > &echo()
> > >
> > > produce the exact same result & log
> > >
> > > :(
> > >
> > >> * I think you need to '' the sofia uri /b
> > >
> > >
>
>

-- 
Erick Johnson

sip/email erick at junctionnetworks.com
1-800-801-3381 x7006
Software Engineer
Junction Networks




From gilbertandrew at me.com  Tue Dec  9 12:54:01 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 09 Dec 2008 15:54:01 -0500
Subject: [Freeswitch-users] Question Regarding ASR/TTS and CMU OSS Projects
Message-ID: <BFC0BBD3-D9A7-48DA-A8CD-E8448B75A9F2@me.com>

Curious if anyone has practical real world input on training CMU based  
ASR engines (Sphinx, PocketSphinx) and / or creating and tuning voices  
for the TTS related components.

Just trying to understand how hard it is, what the realistic gap is to  
use these tools in real world applications.





From frank at impactfax.com  Tue Dec  9 14:35:47 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 9 Dec 2008 17:35:47 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <4CF7C22E-64D4-4EEF-8153-4999E06E3A9F@me.com>
Message-ID: <09aa01c95a4e$7b7a6ea0$33014c0a@ws4>

On our last bill, the carrier said we had 27% short duration calls
(maybe they are wrong but it was on the bill).  It is definitely not
call center. But these callers hangup as soon as they hear answer
machine or most of the time a ring back tone from cell phone.  This
class of caller will call a cell phone, hear the ring back, hangup right
away and then call back another 2 minutes later and repeat the cycle.
 
So, if I have to make it work the way I suggested (hold the connection
open for at least the minimum time, how might you suggest I do it in the
dial plan?
 
-----Original Message-----
 
Don't want the tone to be wrong here, but this makes no sense. Carriers
surcharge like this precisely to guard against call center, predictive
and other mass outbound calling scenarios.
 
It just doesn't make since, math wise, that individuals hanging up on
voice mail are going to significantly impact overall ACD stats, etc. So
unless you have a very strange set of use cases or are pushing another
category of traffic (ie call center) that skews you overall relationship
with the carrier - I would go back and re-negotiate your arrangement.
 
Yes, FS is a b2bua and all is possible. But it is probably a better use
of time to approach this as a business issue.
 
My 2 cents.
 
 
On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:



How can FS force a Minimum call duration for a FS caller (someone
calling out of FS)? 
 
We have a carrier that penalizes us with a surcharge for short duration
calls (sound familiar?).
 
So when a FS caller (not a call center or predictive dialer) calls a
cell phone and gets a ring tone or calls an answering machine, the FS
caller hangs up because they do not want to leave a message.  But they
do this in less then a few seconds after the call is answered.  This
becomes a short duration call and bang the surcharge applies.  It is
actually cheaper to pay for a longer call time (6 seconds in this case)
and avoid the short duration surcharge.  But the FS caller does not know
this.
 
So, how can FS hold the connection to the called party open for at least
the minimum amount of time I need to avoid the short call charge. even
though my FS caller has already hung up the phone on his end?  I would
like to do this in the xml dialplanif possible.
 
Thanks
 
-Frank
 
 
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From msc at freeswitch.org  Tue Dec  9 15:53:39 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Dec 2008 15:53:39 -0800
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <09aa01c95a4e$7b7a6ea0$33014c0a@ws4>
References: <4CF7C22E-64D4-4EEF-8153-4999E06E3A9F@me.com>
	<09aa01c95a4e$7b7a6ea0$33014c0a@ws4>
Message-ID: <87f2f3b90812091553s561b6b48kda080b1bee0dd775@mail.gmail.com>

Can you paste your dialplan entry here? I have some thoughts but it
would be better if I knew what you were doing before I go any further.
-MC



On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <frank at impactfax.com> wrote:
> On our last bill, the carrier said we had 27% short duration calls (maybe
> they are wrong but it was on the bill).  It is definitely not call center.
> But these callers hangup as soon as they hear answer machine or most of the
> time a ring back tone from cell phone.  This class of caller will call a
> cell phone, hear the ring back, hangup right away and then call back another
> 2 minutes later and repeat the cycle.
>
>
>
> So, if I have to make it work the way I suggested (hold the connection open
> for at least the minimum time, how might you suggest I do it in the dial
> plan?
>
>
>
> -----Original Message-----
>
>
> Don't want the tone to be wrong here, but this makes no sense. Carriers
> surcharge like this precisely to guard against call center, predictive and
> other mass outbound calling scenarios.
>
>
>
> It just doesn't make since, math wise, that individuals hanging up on voice
> mail are going to significantly impact overall ACD stats, etc. So unless you
> have a very strange set of use cases or are pushing another category of
> traffic (ie call center) that skews you overall relationship with the
> carrier - I would go back and re-negotiate your arrangement.
>
>
>
> Yes, FS is a b2bua and all is possible. But it is probably a better use of
> time to approach this as a business issue.
>
>
>
> My 2 cents.
>
>
>
>
>
> On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:
>
> How can FS force a Minimum call duration for a FS caller (someone calling
> out of FS)?
>
>
>
> We have a carrier that penalizes us with a surcharge for short duration
> calls (sound familiar?).
>
>
>
> So when a FS caller (not a call center or predictive dialer) calls a cell
> phone and gets a ring tone or calls an answering machine, the FS caller
> hangs up because they do not want to leave a message.  But they do this in
> less then a few seconds after the call is answered.  This becomes a short
> duration call and bang the surcharge applies.  It is actually cheaper to pay
> for a longer call time (6 seconds in this case) and avoid the short duration
> surcharge.  But the FS caller does not know this.
>
>
>
> So, how can FS hold the connection to the called party open for at least the
> minimum amount of time I need to avoid the short call charge? even though my
> FS caller has already hung up the phone on his end?  I would like to do this
> in the xml dialplanif possible.
>
>
>
> Thanks
>
>
>
> -Frank
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From gilbertandrew at me.com  Tue Dec  9 16:27:36 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Tue, 09 Dec 2008 19:27:36 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <09aa01c95a4e$7b7a6ea0$33014c0a@ws4>
References: <09aa01c95a4e$7b7a6ea0$33014c0a@ws4>
Message-ID: <D149AA89-71BB-4626-918D-9B1845EBEC9B@me.com>

What do your records say? Ie do they balance to what the carrier  
claims? You should at a minimum have macro level data to confirm  
against.

27% seems high, but even at that level if you assume your remaining  
population is "normal" you are still no where close to call center /  
predictive traffic in the overall sense. For example, 2 minutes ACD on  
the normal population is still almost 90 seconds overall. Compare this  
to outbound call centers that might have an overall ACD in the 10-30  
second range and have well over 50%, probably much higher, short  
duration. I would tell your carrier to stop being silly, or find  
another one.

I am unsure you can do it just in the dialplan, but it is a somewhat  
trivial app. The issue is it is difficult to safely avoid scenarios  
where leg B might actually be a real person, talking to dead air. This  
is not good citizenship. It breaks implicit assumptions about network  
behavior and is unfair to end users. It is illegal if applied to a  
predictive scenario.


On Dec 9, 2008, at 5:35 PM, Frank @ Impact wrote:

> On our last bill, the carrier said we had 27% short duration calls  
> (maybe they are wrong but it was on the bill).  It is definitely not  
> call center. But these callers hangup as soon as they hear answer  
> machine or most of the time a ring back tone from cell phone.  This  
> class of caller will call a cell phone, hear the ring back, hangup  
> right away and then call back another 2 minutes later and repeat the  
> cycle.
>
> So, if I have to make it work the way I suggested (hold the  
> connection open for at least the minimum time, how might you suggest  
> I do it in the dial plan?
>
> -----Original Message-----
>
> Don't want the tone to be wrong here, but this makes no sense.  
> Carriers surcharge like this precisely to guard against call center,  
> predictive and other mass outbound calling scenarios.
>
> It just doesn't make since, math wise, that individuals hanging up  
> on voice mail are going to significantly impact overall ACD stats,  
> etc. So unless you have a very strange set of use cases or are  
> pushing another category of traffic (ie call center) that skews you  
> overall relationship with the carrier - I would go back and re- 
> negotiate your arrangement.
>
> Yes, FS is a b2bua and all is possible. But it is probably a better  
> use of time to approach this as a business issue.
>
> My 2 cents.
>
>
> On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:
>
>
> How can FS force a Minimum call duration for a FS caller (someone  
> calling out of FS)?
>
> We have a carrier that penalizes us with a surcharge for short  
> duration calls (sound familiar?).
>
> So when a FS caller (not a call center or predictive dialer) calls a  
> cell phone and gets a ring tone or calls an answering machine, the  
> FS caller hangs up because they do not want to leave a message.  But  
> they do this in less then a few seconds after the call is answered.   
> This becomes a short duration call and bang the surcharge applies.   
> It is actually cheaper to pay for a longer call time (6 seconds in  
> this case) and avoid the short duration surcharge.  But the FS  
> caller does not know this.
>
> So, how can FS hold the connection to the called party open for at  
> least the minimum amount of time I need to avoid the short call  
> charge? even though my FS caller has already hung up the phone on  
> his end?  I would like to do this in the xml dialplanif possible.
>
> Thanks
>
> -Frank
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frank at impactfax.com  Tue Dec  9 17:10:41 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 9 Dec 2008 20:10:41 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <87f2f3b90812091553s561b6b48kda080b1bee0dd775@mail.gmail.com>
Message-ID: <0a4701c95a64$1ff68990$33014c0a@ws4>

Pretty simple...

<?xml version="1.0"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="Regex/XML Dialplan">
    <context name="default">
      <extension name="myextension">
        <condition>
          <action application="export"
data="effective_caller_id_name="/>
          <action application="export"
data="effective_caller_id_number=5555551212"/>
	  <action application="enum" data="5555552121"/>
	  <action application="bridge" data="${enum_auto_route}"/> 
	  <action application="hangup"/>
        </condition>
      </extension>
    </context>
  </section>
</document>

-----Original Message-----
 
Can you paste your dialplan entry here? I have some thoughts but it
would be better if I knew what you were doing before I go any further.
-MC



On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <frank at impactfax.com>
wrote:
> On our last bill, the carrier said we had 27% short duration calls
(maybe
> they are wrong but it was on the bill).  It is definitely not call
center.
> But these callers hangup as soon as they hear answer machine or most
of the
> time a ring back tone from cell phone.  This class of caller will call
a
> cell phone, hear the ring back, hangup right away and then call back
another
> 2 minutes later and repeat the cycle.
>
>
>
> So, if I have to make it work the way I suggested (hold the connection
open
> for at least the minimum time, how might you suggest I do it in the
dial
> plan?
>
>
>
> -----Original Message-----
>
>
> Don't want the tone to be wrong here, but this makes no sense.
Carriers
> surcharge like this precisely to guard against call center, predictive
and
> other mass outbound calling scenarios.
>
>
>
> It just doesn't make since, math wise, that individuals hanging up on
voice
> mail are going to significantly impact overall ACD stats, etc. So
unless you
> have a very strange set of use cases or are pushing another category
of
> traffic (ie call center) that skews you overall relationship with the
> carrier - I would go back and re-negotiate your arrangement.
>
>
>
> Yes, FS is a b2bua and all is possible. But it is probably a better
use of
> time to approach this as a business issue.
>
>
>
> My 2 cents.
>
>
>
>
>
> On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote:
>
> How can FS force a Minimum call duration for a FS caller (someone
calling
> out of FS)?
>
>
>
> We have a carrier that penalizes us with a surcharge for short
duration
> calls (sound familiar?).
>
>
>
> So when a FS caller (not a call center or predictive dialer) calls a
cell
> phone and gets a ring tone or calls an answering machine, the FS
caller
> hangs up because they do not want to leave a message.  But they do
this in
> less then a few seconds after the call is answered.  This becomes a
short
> duration call and bang the surcharge applies.  It is actually cheaper
to pay
> for a longer call time (6 seconds in this case) and avoid the short
duration
> surcharge.  But the FS caller does not know this.
>
>
>
> So, how can FS hold the connection to the called party open for at
least the
> minimum amount of time I need to avoid the short call charge. even
though my
> FS caller has already hung up the phone on his end?  I would like to
do this
> in the xml dialplanif possible.
>
>
>
> Thanks
>
>
>
> -Frank
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From ack at telefonica.net  Tue Dec  9 18:10:10 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Wed, 10 Dec 2008 03:10:10 +0100
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <191c3a030812040734s4f514f42s9a30a48c93709fd5@mail.gmail.com>
References: <1228352588.25709.42.camel@develop4>
	<191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
	<1228365981.25709.60.camel@develop4>
	<191c3a030812040734s4f514f42s9a30a48c93709fd5@mail.gmail.com>
Message-ID: <1228875010.2477.67.camel@develop4>

Thanks Anthony , you did a great work ! this is fixed in svn r10691.

Some notes for people using Sonus and L3 as was my case :

in var.xml in some scenario you may need :

<X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/>

in sip_profiles/internal.xml :

<param name="rtp-rewrite-timestamps" value="true"/>

might help for some people with rtp issues :

<param name="rtp-timer-name" value="none"/>

If you have issues with DTMF and timestamps add also :

<param name="pass-rfc2833" value="true"/>

I've a little issues with DTMF from VOIP , i i'll figure out can could
be the issue , from PSTN all works like a charm :)

Cheers,

El jue, 04-12-2008 a las 09:34 -0600, Anthony Minessale escribi?:
> most likely it's because during the time you are dong artificial
> ringback the other side is not doing RTP right.
> 
> When the call is answered we flush the rtp buffer and your missing
> audio is probably flushed with it.
> so you can choose to have a 3 second delay or erase the 3 seconds as
> it does now.
> 
> This is a known problem with sonus which has been proven to build up
> an audio delay during the time
> you are waiting for the call to answer.  I'm sure you prefer the way
> it is to a large audio delay.
> 
> 
> 
> On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <ack at telefonica.net>
> wrote:
>         No TDM , all is SIP :
>         
>         
>         PSTN ---> Sip Proxy_A --> FS ( brigde )
>         ringback/transfer_ringback
>         -> Sip Proxy_B --> PSTN
>         
>         
>         In logfile i think you can get some details about Media
>         Gateways
>         ( Sonus ) PSTN inbound / outbound is provided by Level3.
>         
>         I can get a capture of a call if you want, in capture the
>         audio is not
>         missing, issue with :
>         
>         - rtp buffer ?
>         - Sonus ?
>         
>         Let me know anything you need so i can provide a log or create
>         a new
>         scenario.
>         
>         
>         Thanks,
>         
>         El mi?, 03-12-2008 a las 22:12 -0600, Anthony Minessale
>         escribi?:
>         
>         > what does PSTN represent?
>         >
>         > I know what the PSTN is but how are you reaching it?
>         > is it TDM, SIP etc... what gateway type other details.
>         >
>         >
>         > On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero
>         <ack at telefonica.net>
>         > wrote:
>         >         Hi guys,
>         >
>         >          I've a strange issue with FS , version svn
>         -r10584 ,
>         >         when FS bridges a call first 3 seconds of audio are
>         missing ,
>         >         looks that
>         >         only happens on PSTN calls and using ringback or
>         >         transfer_ringback. This
>         >         only happens in calls from PSTN , not from VOIP.
>         Some
>         >         scenarios i tried
>         >         to isolate this issue :
>         >
>         >
>         >         - Issue
>         >
>         >         PSTN --> FS ( brigde ) ringback/transfer_ringback ->
>         PSTN
>         >
>         >         - Good setting bypass_media before run bridge but i
>         need rtp
>         >         in FS path
>         >
>         >         PSTN --> FS ( brigde ) ringback/transfer_ringback ->
>         PSTN
>         >
>         >         - Good
>         >
>         >         PSTN --> FS ( brigde ) WITHOUT
>         ringback/transfer_ringback ->
>         >         PSTN
>         >
>         >         - Always good
>         >
>         >         VOIP --> FS ( brigde ) -> PSTN
>         >
>         >
>         >         Dialplan has nothing wrong ( i guess ):
>         >
>         >         <extension name="Transfers">
>         >            <condition field="destination_number"
>         >         expression="^1??XXXXXXXXXX$">
>         >              <action application="answer"/>
>         >              <action application="speak" data="cepstral|
>         Allison-8kHz|
>         >         blah"/>
>         >              <action application="set"
>         >         data="hangup_after_bridge=false"/>
>         >              <action application="set"
>         data="playback_terminators=#"/>
>         >              <action application="set" data="ringback=
>         $${us-ring}"/>
>         >              <action application="set"
>         data="transfer_ringback=
>         >         $${hold_music}"/>
>         >              <action application="set"
>         data="effective_caller_id_name=
>         >         ${caller_id_name}"/>
>         >              <action application="set"
>         >         data="effective_caller_id_number=
>         >         ${caller_id_number}"/>
>         >              <action application="set"
>         data="originate_timeout=30"/>
>         >              <action application="set"
>         data="call_timeout=30"/>
>         >              <action application="bridge"
>         >         data="sofia/default/18008226235 at PSTN_GW"/>
>         >              <action application="speak" data="cepstral|
>         Allison-8kHz|
>         >         Transfer
>         >         finished"/>
>         >              <action application="hangup"/>
>         >            </condition>
>         >          </extension>
>         >
>         >
>         >
>         >         Any ideas ?
>         >
>         >         Attached log of FS ( F8 from console ).
>         >
>         >
>         >         Thanks in advance !
>         >
>         >         --
>         >         Angel Carpintero
>         >         ack ( at ) telefonica ( dot ) net
>         >
>         >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF
>         AC2C CA61
>         >         6EF1 B90D
>         >
>         >
>         >
>         
>         > --
>         > Anthony Minessale II
>         >
>         > FreeSWITCH http://www.freeswitch.org/
>         > ClueCon http://www.cluecon.com/
>         >
>         > AIM: anthm
>         > MSN:anthony_minessale at hotmail.com
>         > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>         > IRC: irc.freenode.net #freeswitch
>         >
>         > FreeSWITCH Developer Conference
>         > sip:888 at conference.freeswitch.org
>         > iax:guest at conference.freeswitch.org/888
>         > googletalk:conf+888 at conference.freeswitch.org
>         > pstn:213-799-1400
>         
>         
>         --
>         
>         Angel Carpintero
>         ack ( at ) telefonica ( dot ) net
>         
>         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
>         6EF1 B90D
>         
>         
>         
>         _______________________________________________
>         Freeswitch-users mailing list
>         Freeswitch-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

"No basta saber, hay que aplicar lo que se sabe; 
no basta querer hacerlas cosas, hay que hacerlas".

"Knowing is not enough; we must apply. 
 Willing is not enough; we must do"

 Johann Wolfgang von Goethe
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From jpalley at idapted.com  Tue Dec  9 18:58:38 2008
From: jpalley at idapted.com (Jonathan Palley)
Date: Wed, 10 Dec 2008 10:58:38 +0800
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <B7BC254D-855D-4247-82B9-BEF52C024E9E@jerris.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>
	<B7BC254D-855D-4247-82B9-BEF52C024E9E@jerris.com>
Message-ID: <2d8777c00812091858s1c5fa3d9m4e5f11163b894628@mail.gmail.com>

I can offer a bit of a bounty for this.  Can anyone else chip in?

Thanks -
JP

On Tue, Dec 9, 2008 at 11:45 PM, Michael Jerris <mike at jerris.com> wrote:

> It is something we have been discussing as we need these stats to do
> rtcp properly but we have not written any code to do so.  It is
> "somewhat" difficult.  I would say it is on our minds but not on any
> roadmap just yet.
>
> MIke
>
> On Dec 8, 2008, at 11:37 PM, Jonathan Palley wrote:
>
> > I'm curious to start a discussion on being able to query a channel
> > and get statistics on the incoming jitter and packet loss
> > (calculated from the RTP, not RTCP).
> >
> > Is this on the roadmap?  Is it hard to do?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jonathan Palley | Idapted Inc.
jpalley at idapted.com
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From c_cav_01 at yahoo.com  Tue Dec  9 20:24:29 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 9 Dec 2008 20:24:29 -0800 (PST)
Subject: [Freeswitch-users]  incoming call routing <domain>
Message-ID: <20928933.post@talk.nabble.com>


Cable modem <----> nat router <---->  fs

fs is set as DMZ on nat router so all packets get there.

My ipv4 address is 192.168.0.x  The nat router holds the public IP.  Public
IP is a registered domain sparkz.tv so addressable from the internet cloud. 
Since fs is DMZ, all requests for sparkz.tv or sip.sparkz.tv are resolved
and so IP routing is good.

So I'm trying to get external sip/soft phones registered and routed
properly.  The domain/server set in the phone client is sip.sparkz.tv:5080,
since the wiki says they need to be set that way???

I have created a conf/directory/sip.sparkz.tv.xml  and a
conf/directory/sip.sparkz.tv where I have users registration info.

conf/directory/sip.sparkz.tv.xml was copied from default.xml and has:
      
    </params>


I have modified conf/sip_profiles/external.xml and added an <alias
name="sip.sparkz.tv"/>

External phones are registering and are visible under sofia status profiles
external and sip.sparkz.tv

Calls outbound from the phones are being routed properly.

Calls inbound to their DID's are not.
Calls to softphones on the local private net 192.168.0.x register and route
properly.
vars.xml sets domain to ip_v4...
the default.xml dialplan seems to iif the profile to either nat or default.. 
so I end up with the call going to DID at 192.168.0.x rather than the
registered interface...

I'm routing the calls in the dialplan to bridge to user/$1@$${domain}  but
$${domain} is set to ip_v4 so it's wrong...

Any clues what I need to do next to get them routing properly?  I want to be
able to support multiple domains. how do I do this correctly?
-- 
View this message in context: http://www.nabble.com/incoming-call-routing-%3Cdomain%3E-tp20928933p20928933.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From c_cav_01 at yahoo.com  Tue Dec  9 20:25:32 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Tue, 9 Dec 2008 20:25:32 -0800 (PST)
Subject: [Freeswitch-users]  incoming call routing <domain>
Message-ID: <20928933.post@talk.nabble.com>


Cable modem <----> nat router <---->  fs

fs is set as DMZ on nat router so all packets get there.

My ipv4 address is 192.168.0.x  The nat router holds the public IP.  Public
IP is a registered domain sparkz.tv so addressable from the internet cloud. 
Since fs is DMZ, all requests for sparkz.tv or sip.sparkz.tv are resolved
and so IP routing is good.

So I'm trying to get external sip/soft phones registered and routed
properly.  The domain/server set in the phone client is sip.sparkz.tv:5080,
since the wiki says they need to be set that way???

I have created a conf/directory/sip.sparkz.tv.xml  and a
conf/directory/sip.sparkz.tv where I have users registration info.

conf/directory/sip.sparkz.tv.xml was copied from default.xml and has:

      param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/${dialed_user}@${dialed_domain})}"
    /params


I have modified conf/sip_profiles/external.xml and added an <alias
name="sip.sparkz.tv"/>

External phones are registering and are visible under sofia status profiles
external and sip.sparkz.tv

Calls outbound from the phones are being routed properly.

Calls inbound to their DID's are not.
Calls to softphones on the local private net 192.168.0.x register and route
properly.
vars.xml sets domain to ip_v4...
the default.xml dialplan seems to iif the profile to either nat or default.. 
so I end up with the call going to DID at 192.168.0.x rather than the
registered interface...

I'm routing the calls in the dialplan to bridge to user/$1@$${domain}  but
$${domain} is set to ip_v4 so it's wrong...

Any clues what I need to do next to get them routing properly?  I want to be
able to support multiple domains. how do I do this correctly?
-- 
View this message in context: http://www.nabble.com/incoming-call-routing-%3Cdomain%3E-tp20928933p20928933.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From dave at 3c.co.uk  Tue Dec  9 21:26:27 2008
From: dave at 3c.co.uk (David Knell)
Date: Wed, 10 Dec 2008 05:26:27 +0000
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <493ED9E6.3000803@junctionnetworks.com>
References: <493ED9E6.3000803@junctionnetworks.com>
Message-ID: <493F5303.30208@3c.co.uk>

Hi Erick,

Not sure if you've tried this (or if it'll help), but you can force 
routing in the dialplan like so:
<action application="set" data="sip_h_Route=<sip:@11.22.33.44;lr>" />
<action application="bridge" data="sofia/gateway/gw/$1"/>

Cheers --

Dave


> i forgot to give you the pastebin URL
> http://pastebin.freeswitch.org/6379
>
>   
>> I'm running latest trunk - Revision: 10682
>>
>> I've been doing an ngrep on my external freeswitch SIP port and FS
>> is not sending any SIP packets anywhere when I run the following command.
>> Bumping up TPORT_LOG to 9 also confirms this, as no SIP packets are 
>> logged.
>>
>> originate
>> 'sofia/external/erick at ejjohnson.org;fs_path=proxybeta.jnctn.net' 
>> &echo()
>>
>> Also, just to be clear, when I remove ";fs_path=..." from the command
>> above a call
>> is set up normally to erick at ejjohnson.org and the SIP packets are 
>> logged
>> to console.
>>
>> Thanks guys.
>>
>>     
>>> What SVN rev are you running? Also, could you do a SIP trace?
>>> TPORT_LOG=1 && /usr/local/freeswitch/bin/freeswitch
>>> Pastebin the output of that and we'll take it from there.
>>> -MC
>>>
>>> On Tue, Dec 9, 2008 at 11:41 AM, Erick Johnson
>>> <erick at junctionnetworks.com> wrote:
>>>       
>>>> Both:
>>>>
>>>> originate sofia/external/'erick at
>>>>         
>>> ejjohnson.org;fs_path=proxybeta.foo.net'
>>>       
>>>> &echo()
>>>> originate sofia/external/erick at
>>>>         
>>> ejjohnson.org;fs_path=proxybeta.foo.net
>>>       
>>>> &echo()
>>>>
>>>> produce the exact same result & log
>>>>
>>>> :(
>>>>
>>>>         
>>>>> * I think you need to '' the sofia uri /b
>>>>>           
>>>>         
>>     
>
>   


-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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From helmut.kuper at ewetel.de  Tue Dec  9 23:35:36 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 10 Dec 2008 08:35:36 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>
	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>
Message-ID: <493F7148.40705@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

on my ubuntu 8.04 I have libtiff4 and libtiff4-dev installed. libtiff
and libtiff-dev is not installed. I gonna test it today

regards
helmut


Michael Collins schrieb:
> Helmut,
> I think Mike J was pointing out that spandsp needs libtiff and
> libtiff-devel in order to compile, so you need to do that first and
> then compile freeswitch.
> -MC
> 
> On Tue, Dec 9, 2008 at 8:09 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> Hi Michael,
> 
> don't know if you get me right: Everything is there, but obviously FS
> makefile has to compile "libs/spandsp/src" before mod_fax (at least I
> guess so). Currently the Makefile referred to libspandsp.la before it is
> compiled.
> 
> regards
> helmut
> 
> 
> 
> Michael Jerris schrieb:
>>>> make sure you have libtiff and libtiff dev packages installed then re-
>>>> configure freeswitch
>>>>
>>>> Mike
>>>>
>>>> On Dec 9, 2008, at 5:07 AM, Helmut Kuper wrote:
>>>>
>>>> Hello,
>>>>
>>>> I tried to compile mod_fax today with trunk from yesterday. A 'make'
>>>> in
>>>> FS trunk directory led to an error saying that libspandsp.la wasn't
>>>> found in libs/spandsp/src. So I had to configure and compile (make)
>>>> spandsp manually before compiling FS.
>>>>
>>>> regards
>>>> helmut
>>>>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
> 
>>
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
>>

> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From jalsot at gmail.com  Wed Dec 10 00:23:26 2008
From: jalsot at gmail.com (Tamas)
Date: Wed, 10 Dec 2008 09:23:26 +0100
Subject: [Freeswitch-users] Jitter + Packet Loss
In-Reply-To: <2d8777c00812091858s1c5fa3d9m4e5f11163b894628@mail.gmail.com>
References: <2d8777c00812082037q5b5cbd33mc1a177e3d51a8993@mail.gmail.com>	<B7BC254D-855D-4247-82B9-BEF52C024E9E@jerris.com>
	<2d8777c00812091858s1c5fa3d9m4e5f11163b894628@mail.gmail.com>
Message-ID: <493F7C7E.6030805@gmail.com>

Hello,

I've added bounty for RTCP already:
http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support

I know that the requester wants stats for RTP but maybe we could make a
joint bounty ;)

Regards,
    Tamas

Jonathan Palley ?rta:
> I can offer a bit of a bounty for this.  Can anyone else chip in?
>
> Thanks -
> JP
>
> On Tue, Dec 9, 2008 at 11:45 PM, Michael Jerris <mike at jerris.com
> <mailto:mike at jerris.com>> wrote:
>
>     It is something we have been discussing as we need these stats to do
>     rtcp properly but we have not written any code to do so.  It is
>     "somewhat" difficult.  I would say it is on our minds but not on any
>     roadmap just yet.
>
>     MIke
>
>     On Dec 8, 2008, at 11:37 PM, Jonathan Palley wrote:
>
>     > I'm curious to start a discussion on being able to query a channel
>     > and get statistics on the incoming jitter and packet loss
>     > (calculated from the RTP, not RTCP).
>     >
>     > Is this on the roadmap?  Is it hard to do?
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Jonathan Palley | Idapted Inc.
> jpalley at idapted.com <mailto:jpalley at idapted.com>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From brian at freeswitch.org  Wed Dec 10 01:45:34 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 03:45:34 -0600
Subject: [Freeswitch-users] incoming call routing <domain>
In-Reply-To: <20928933.post@talk.nabble.com>
References: <20928933.post@talk.nabble.com>
Message-ID: <498842E3-2204-4204-AF40-43AA344E0C35@freeswitch.org>

Join IRC so you can interact with people real time.  Your setup  
require a deep understanding of SIP and FreeSWITCH to setup correctly.

/b

On Dec 9, 2008, at 10:25 PM, ccav wrote:

>
> Cable modem <----> nat router <---->  fs
>
> fs is set as DMZ on nat router so all packets get there.
>
> My ipv4 address is 192.168.0.x  The nat router holds the public IP.   
> Public
> IP is a registered domain sparkz.tv so addressable from the internet  
> cloud.
> Since fs is DMZ, all requests for sparkz.tv or sip.sparkz.tv are  
> resolved
> and so IP routing is good.
>
> So I'm trying to get external sip/soft phones registered and routed
> properly.  The domain/server set in the phone client is  
> sip.sparkz.tv:5080,
> since the wiki says they need to be set that way???
>
> I have created a conf/directory/sip.sparkz.tv.xml  and a
> conf/directory/sip.sparkz.tv where I have users registration info.
>
> conf/directory/sip.sparkz.tv.xml was copied from default.xml and has:
>
>      param name="dial-string"
> value="{presence_id=${dialed_user}@$ 
> {dialed_domain},transfer_fallback_extension=${dialed_user}}$ 
> {sofia_contact(${dialed_domain}/${dialed_user}@${dialed_domain})}"
>    /params
>
>
> I have modified conf/sip_profiles/external.xml and added an <alias
> name="sip.sparkz.tv"/>
>
> External phones are registering and are visible under sofia status  
> profiles
> external and sip.sparkz.tv
>
> Calls outbound from the phones are being routed properly.
>
> Calls inbound to their DID's are not.
> Calls to softphones on the local private net 192.168.0.x register  
> and route
> properly.
> vars.xml sets domain to ip_v4...
> the default.xml dialplan seems to iif the profile to either nat or  
> default..
> so I end up with the call going to DID at 192.168.0.x rather than the
> registered interface...
>
> I'm routing the calls in the dialplan to bridge to user/$1@$$ 
> {domain}  but
> $${domain} is set to ip_v4 so it's wrong...
>
> Any clues what I need to do next to get them routing properly?  I  
> want to be
> able to support multiple domains. how do I do this correctly?
> -- 
> View this message in context: http://www.nabble.com/incoming-call-routing-%3Cdomain%3E-tp20928933p20928933.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From carole.olivier at enst.fr  Wed Dec 10 05:37:05 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 10 Dec 2008 05:37:05 -0800 (PST)
Subject: [Freeswitch-users]  segmentation fault by auto record
Message-ID: <20935513.post@talk.nabble.com>


Hello,

I have intalled Freeswitch from opensuse.org as a rpm. I have opensuse 10.3.
I did not make any big configuration, I have just changed a little the
default dialplan and adapted some other files like conference.conf.xml.

I have created a new profile in conference.conf.xml and add the command to
order the automatic record of the conferences:

<profile name="profilespeaker">
   
   
   
   
   
   
   
   
   
</profile>

I have the following extension in my dialplan that uses this profile:

<extension name="call_speakers">
 <condition field="destination_number" expression="^0911$">
   <action application="set"
data="conference_auto_outcall_caller_id_name=call_speakers" /> 
   <action application="set" data="conference_auto_outcall_id_number=0911"
/> 
   <action application="set" data="conference_auto_outcall_timeout=60" /> 
   <action application="set" data="conference_auto_outcall_flags=mute" /> 
   <action application="set" data="api_hangup_hook=conference conf_speaker
kick all" /> 
<!--  devices called  --> 
  <action application="conference_set_auto_outcall"
data="user/1021@$${domain}" /> 
  <action application="conference_set_auto_outcall"
data="user/1022@$${domain}" /> 
<!-- creation conference --> 
  <action application="conference" data="conf_speaker at profilespeaker" /> 
 </condition>
</extension>

1021 and 1022 are IP loud speakers.
I call them with 0911, they answer, I can talk and everything works well.
However, when I hang up a segmentation fault appears and freeswitch
shutdowns. In recordings I can find a file which corresponds to the recorded
call but this is empty. Nothing has been recorded.
If I comment the line with the recording command then it works without
problem except the recording...

I have joined two files: one contains the errors that appears when I run
freeswitch and the other what happens if I call the extension 0911.

http://www.nabble.com/file/p20935513/running_freeswitch.txt
running_freeswitch.txt 
http://www.nabble.com/file/p20935513/call_extension.txt call_extension.txt 

If someone has an idea, it would be very helpful
Thanks!!
Carole
-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20935513.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From carole.olivier at enst.fr  Wed Dec 10 05:50:09 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 10 Dec 2008 05:50:09 -0800 (PST)
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20935513.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
Message-ID: <20935752.post@talk.nabble.com>


Sorry, here is the profile profile speaker

<profile name="profilespeaker">
  < param name="rate" value="8000" /> 
  < param name="interval" value="20" /> 
  < param name="energy-level" value="300" /> 
  < param name="caller-id-name" value="$${outbound_caller_name}" /> 
  < param name="caller-id-number" value="$${outbound_caller_id}" /> 
  < param name="comfort-noise-level" value="1400" /> 
  < param name="comfort-noise" value="true" /> 
  < param name="member_flags" value="waste" /> 
  < param name="auto-record"
value="$${base_dir}/recordings/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"
/> 
  </profile>

-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20935752.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From john at loopfx.com  Wed Dec 10 06:36:33 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 09:36:33 -0500
Subject: [Freeswitch-users] No audio after transfer
Message-ID: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>

Sorry to repost, but I haven't heard anything back on this in a little
while.

 

I checked out the trunk last week.  I'm on revision 10597.  

 

Thanks,

John

 

From: John Rutherford 
Sent: Monday, December 08, 2008 4:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: No audio after transfer

 

I'm trying to get an attended transfer work with freeSWITCH, but it's
not quite working.  I have Microsoft Speech Server on one side and
Televantage on the other.  

 

MSS is originating a call, which freeSWITCH is bridging to Televantage.
That calls connects just fine.  Then, MSS sends a re-INVITE to
Televantage to put the call on hold.  This works too.  Then, MSS
originates another call to freeSWITCH, which is again bridged to
Televantage.  This works fine too.  

 

Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
should be complete, but there is no audio between the two calls-just
silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not
seeing anything obviously wrong.  

 

After the REFER, I can see audio for both calls going between freeSWITCH
and Televantage, so it seems that the only thing missing is freeSWITCH
routing the audio from one call to the other call and vice-versa.

 

 

Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this.  

 

Thanks!

John

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From frank at impactfax.com  Wed Dec 10 06:46:01 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Wed, 10 Dec 2008 09:46:01 -0500
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <87f2f3b90812091553s561b6b48kda080b1bee0dd775@mail.gmail.com>
Message-ID: <0cd801c95ad6$055cf890$33014c0a@ws4>

This is a bit beyond me, but in the xml dialplan could we use the
execute_on_answer to transfer to an extension that sets up an intercept
of the bleg on hangup_after_bridge?  Or use the api_hangup_hook to
transfer the bleg to another extension after the aleg hangs up?

I have been reading all the wiki information I can and these smell like
they might help.  But it is not clear to me how they would be pieced
together exactly to achieve this end.

MC, any thoughts?

-----Original Message-----
 
Can you paste your dialplan entry here? I have some thoughts but it
would be better if I knew what you were doing before I go any further.
-MC



On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <frank at impactfax.com>
wrote:
> On our last bill, the carrier said we had 27% short duration calls
(maybe
> they are wrong but it was on the bill).  It is definitely not call
center.
> But these callers hangup as soon as they hear answer machine or most
of the
> time a ring back tone from cell phone.  This class of caller will call
a
> cell phone, hear the ring back, hangup right away and then call back
another
> 2 minutes later and repeat the cycle.
>
>





From anthony.minessale at gmail.com  Wed Dec 10 07:34:27 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Dec 2008 09:34:27 -0600
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
Message-ID: <191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>

Are you expecting every message that MSS sends FS to be in turn sent to
televantage?
That is a proxy behaviour.  Since FreeSWITCH plays the role of a b2bua it
will not pass the
messages across a bridge.



On Wed, Dec 10, 2008 at 8:36 AM, John Rutherford <john at loopfx.com> wrote:

>  Sorry to repost, but I haven't heard anything back on this in a little
> while.
>
>
>
> I checked out the trunk last week.  I'm on revision 10597.
>
>
>
> Thanks,
>
> John
>
>
>
> *From:* John Rutherford
> *Sent:* Monday, December 08, 2008 4:36 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* No audio after transfer
>
>
>
> I'm trying to get an attended transfer work with freeSWITCH, but it's not
> quite working.  I have Microsoft Speech Server on one side and Televantage
> on the other.
>
>
>
> MSS is originating a call, which freeSWITCH is bridging to Televantage.
> That calls connects just fine.  Then, MSS sends a re-INVITE to Televantage
> to put the call on hold.  This works too.  Then, MSS originates another call
> to freeSWITCH, which is again bridged to Televantage.  This works fine too.
>
>
>
>
> Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
> should be complete, but there is no audio between the two calls?just
> silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not seeing
> anything obviously wrong.
>
>
>
> After the REFER, I can see audio for both calls going between freeSWITCH
> and Televantage, so it seems that the only thing missing is freeSWITCH
> routing the audio from one call to the other call and vice-versa.
>
>
>
>
>
> Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
> logs, and I can easily reproduce this.
>
>
>
> Thanks!
>
> John
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Wed Dec 10 07:41:52 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 07:41:52 -0800
Subject: [Freeswitch-users] how to force a MINIMUM call duration
In-Reply-To: <0cd801c95ad6$055cf890$33014c0a@ws4>
References: <87f2f3b90812091553s561b6b48kda080b1bee0dd775@mail.gmail.com>
	<0cd801c95ad6$055cf890$33014c0a@ws4>
Message-ID: <87f2f3b90812100741r2067549fmc88674cf99af7b7e@mail.gmail.com>

On Wed, Dec 10, 2008 at 6:46 AM, Frank @ Impact <frank at impactfax.com> wrote:
> This is a bit beyond me, but in the xml dialplan could we use the
> execute_on_answer to transfer to an extension that sets up an intercept
> of the bleg on hangup_after_bridge?  Or use the api_hangup_hook to
> transfer the bleg to another extension after the aleg hangs up?
>
> I have been reading all the wiki information I can and these smell like
> they might help.  But it is not clear to me how they would be pieced
> together exactly to achieve this end.
>
> MC, any thoughts?
>

I spent a fair amount of time playing with these last night but I
didn't find a solution. I'm still thinking about it, but I believe
that most likely it will require some scripting in Lua (or another
language) to be able to get this to work. This is the first time I've
ever dealt with keeping the b-leg alive when the a-leg hangs up - it
is usually the other way around.

I recommend you read up on Lua scripting because you're gonna need it
to be able to pull this off. (You could use a different language but
Lua is the scripting language of choice amongst the FS devs so I
highly recommend using it.)

I will tinker with this a bit later today when I have some more time.
In the meantime if you could start experimenting with the ideas you've
presented and brush up on Lua that would be great. BTW, are you on
IRC?

Thanks,
MC

> -----Original Message-----
>
> Can you paste your dialplan entry here? I have some thoughts but it
> would be better if I knew what you were doing before I go any further.
> -MC
>
>
>
> On Tue, Dec 9, 2008 at 2:35 PM, Frank @ Impact <frank at impactfax.com>
> wrote:
>> On our last bill, the carrier said we had 27% short duration calls
> (maybe
>> they are wrong but it was on the bill).  It is definitely not call
> center.
>> But these callers hangup as soon as they hear answer machine or most
> of the
>> time a ring back tone from cell phone.  This class of caller will call
> a
>> cell phone, hear the ring back, hangup right away and then call back
> another
>> 2 minutes later and repeat the cycle.
>>
>>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From msc at freeswitch.org  Wed Dec 10 07:53:09 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 07:53:09 -0800
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20935513.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
Message-ID: <87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>

Thanks for reporting this. It would be helpful to know a bit more. Can
you start freeswitch and press F12 (or type "version" at the CLI) and
report back what it says?
Also, a backtrace (bt) is generally useful. If you could produce a
"bt" and a "bt full" from you core file that would be extremely
helpful.

see this link for more information:
http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

you should have a "core" file for each segfault that occurred. Use the
gdb program to get the back trace:

gdb /path/to/fs/binary core.xxx

then capture the output from these two commands:

bt <enter>
bt full <enter>

When you type those commands you'll see tons of debugging info;
capture that and put it in a pastebin (pastebin.freeswitch.org) then
report back here.

You can exit the gdb debugger by typing q <enter>

Thanks for helping us collect information!

-MC


On Wed, Dec 10, 2008 at 5:37 AM, Carole O. <carole.olivier at enst.fr> wrote:
>
> Hello,
>
> I have intalled Freeswitch from opensuse.org as a rpm. I have opensuse 10.3.
> I did not make any big configuration, I have just changed a little the
> default dialplan and adapted some other files like conference.conf.xml.
>
> I have created a new profile in conference.conf.xml and add the command to
> order the automatic record of the conferences:
>
> <profile name="profilespeaker">
>
>
>
>
>
>
>
>
>
> </profile>
>
> I have the following extension in my dialplan that uses this profile:
>
> <extension name="call_speakers">
>  <condition field="destination_number" expression="^0911$">
>   <action application="set"
> data="conference_auto_outcall_caller_id_name=call_speakers" />
>   <action application="set" data="conference_auto_outcall_id_number=0911"
> />
>   <action application="set" data="conference_auto_outcall_timeout=60" />
>   <action application="set" data="conference_auto_outcall_flags=mute" />
>   <action application="set" data="api_hangup_hook=conference conf_speaker
> kick all" />
> <!--  devices called  -->
>  <action application="conference_set_auto_outcall"
> data="user/1021@$${domain}" />
>  <action application="conference_set_auto_outcall"
> data="user/1022@$${domain}" />
> <!-- creation conference -->
>  <action application="conference" data="conf_speaker at profilespeaker" />
>  </condition>
> </extension>
>
> 1021 and 1022 are IP loud speakers.
> I call them with 0911, they answer, I can talk and everything works well.
> However, when I hang up a segmentation fault appears and freeswitch
> shutdowns. In recordings I can find a file which corresponds to the recorded
> call but this is empty. Nothing has been recorded.
> If I comment the line with the recording command then it works without
> problem except the recording...
>
> I have joined two files: one contains the errors that appears when I run
> freeswitch and the other what happens if I call the extension 0911.
>
> http://www.nabble.com/file/p20935513/running_freeswitch.txt
> running_freeswitch.txt
> http://www.nabble.com/file/p20935513/call_extension.txt call_extension.txt
>
> If someone has an idea, it would be very helpful
> Thanks!!
> Carole
> --
> View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20935513.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Wed Dec 10 07:58:34 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 09:58:34 -0600
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
Message-ID: <8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>

I have already labbed this up on SVN trunk and I don't get a segfault  
but I get something else that prevents it from working properly.    We  
are working on it today.  Also what version are you running?

/b

On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:

> Thanks for reporting this. It would be helpful to know a bit more. Can
> you start freeswitch and press F12 (or type "version" at the CLI) and
> report back what it says?
> Also, a backtrace (bt) is generally useful. If you could produce a
> "bt" and a "bt full" from you core file that would be extremely
> helpful.
>
> see this link for more information:
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>
> you should have a "core" file for each segfault that occurred. Use the
> gdb program to get the back trace:
>
> gdb /path/to/fs/binary core.xxx
>
> then capture the output from these two commands:
>
> bt <enter>
> bt full <enter>
>
> When you type those commands you'll see tons of debugging info;
> capture that and put it in a pastebin (pastebin.freeswitch.org) then
> report back here.
>
> You can exit the gdb debugger by typing q <enter>
>
> Thanks for helping us collect information!
>
> -MC

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From john at loopfx.com  Wed Dec 10 09:51:59 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 12:51:59 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
	<191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>

No. I realize that's it's a B2BUA and that's exactly what we want.  

 

Everything with the transfer seems to work fine, except that there is no
audio.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Wednesday, December 10, 2008 10:34 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

Are you expecting every message that MSS sends FS to be in turn sent to
televantage?
That is a proxy behaviour.  Since FreeSWITCH plays the role of a b2bua
it will not pass the
messages across a bridge.




On Wed, Dec 10, 2008 at 8:36 AM, John Rutherford <john at loopfx.com>
wrote:

Sorry to repost, but I haven't heard anything back on this in a little
while.

 

I checked out the trunk last week.  I'm on revision 10597.  

 

Thanks,

John

 

From: John Rutherford 
Sent: Monday, December 08, 2008 4:36 PM
To: freeswitch-users at lists.freeswitch.org
Subject: No audio after transfer

 

I'm trying to get an attended transfer work with freeSWITCH, but it's
not quite working.  I have Microsoft Speech Server on one side and
Televantage on the other.  

 

MSS is originating a call, which freeSWITCH is bridging to Televantage.
That calls connects just fine.  Then, MSS sends a re-INVITE to
Televantage to put the call on hold.  This works too.  Then, MSS
originates another call to freeSWITCH, which is again bridged to
Televantage.  This works fine too.  

 

Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
should be complete, but there is no audio between the two calls-just
silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not
seeing anything obviously wrong.  

 

After the REFER, I can see audio for both calls going between freeSWITCH
and Televantage, so it seems that the only thing missing is freeSWITCH
routing the audio from one call to the other call and vice-versa.

 

 

Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this.  

 

Thanks!

John


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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From brian at freeswitch.org  Wed Dec 10 09:58:46 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 11:58:46 -0600
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
	<191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
Message-ID: <BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org>

would be most helpful to capture a pcap of the entire thing by itself  
start to finish.

/b

On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:

> No. I realize that?s it?s a B2BUA and that?s exactly what we want.
>
> Everything with the transfer seems to work fine, except that there  
> is no audio.

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From john at loopfx.com  Wed Dec 10 10:01:56 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 13:01:56 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
	<BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>

I have a pcap, but I'm not able to see anything obviously wrong with it.


 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Wednesday, December 10, 2008 12:59 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

would be most helpful to capture a pcap of the entire thing by itself
start to finish. 

 

/b

 

On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:





No. I realize that's it's a B2BUA and that's exactly what we want. 

 

Everything with the transfer seems to work fine, except that there is no
audio.

 

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From msc at freeswitch.org  Wed Dec 10 10:08:32 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 10:08:32 -0800
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
	<191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
	<BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org>
	<81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>
Message-ID: <87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com>

On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com> wrote:
> I have a pcap, but I'm not able to see anything obviously wrong with it.

We find that some equipment (in fact a lot of equipment) have features
that cause issues to be quite non-obvious, so perhaps you could give
the pcap to Brian for him to review. He's a total ace when it comes to
bug hunting.

-MC

>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Wednesday, December 10, 2008 12:59 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> would be most helpful to capture a pcap of the entire thing by itself start
> to finish.
>
>
>
> /b
>
>
>
> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>
> No. I realize that's it's a B2BUA and that's exactly what we want.
>
>
>
> Everything with the transfer seems to work fine, except that there is no
> audio.
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mszlazak at aol.com  Wed Dec 10 10:39:42 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 13:39:42 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
Message-ID: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>




I?m trying
to route calls from X-lite <--> FS (Nov. 6 2008 svn) <--> Voceo (Prophecy) to use Voxeo?s ASR instead of FS?s built in PocketSphinx/ASR.



All these applications
reside on the same computer/OS (Win XP). I have at Netgear wifi router that
connects a laptop to my desktop in case that matters but I?m not using the
laptop for any of this.



??



I?ve set up
an extension to bridge calls to Voxeo. Here is the entry in file
conf\dialplan\default.xml:



?





??? <extension name="Doctors
Office">



????? <condition
field="destination_number" expression="^2007$">



??????? <action
application="info"/>



??????? <action application="bridge"
data="sofia/internal/sip:Doctorsoffice at 127.0.0.1:5068"/>??????? 



????? </condition>



??? </extension>



?



I hear one
ring then a hang up. No errors in the FS console and Voxeo?s logs ramp up when
I dial the 2007 extension on X-lite but I do not get any audio from the
dialogue script in Voxeo?s Prophecy ASR called ?Doctorsoffice.? 





Any ideas?



?



Thanks.





Mark.




?


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From brian at freeswitch.org  Wed Dec 10 10:45:22 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 12:45:22 -0600
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>
Message-ID: <4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org>

You're a month behind.. I highly recommend you update.  Chances are  
this has already been fixed.


/b

On Dec 10, 2008, at 12:39 PM, mszlazak at aol.com wrote:

> I?m trying to route calls from X-lite <--> FS (Nov. 6 2008 svn) <-->  
> Voceo (Prophecy) to use Voxeo?s ASR instead of FS?s built in  
> PocketSphinx/ASR.
> All these applications reside on the same computer/OS (Win XP). I  
> have at Netgear wifi=2 0router that connects a laptop to my desktop  
> in case that matters but I?m not using the laptop for any of this.
>
> I?ve set up an extension to bridge calls to Voxeo. Here is the entry  
> in file conf\dialplan\default.xml:
>
>     <extension name="Doctors Office">
>       <condition field="destination_number" expression="^2007$">
>         <action application="info"/>
>         <action application="bridge" data="sofia/internal/sip:Doctorsoffice at 127.0.0.1:5068 
> "/>
>       </condition>
>     </extension>
>
> I hear one ring then a hang up. No errors in the FS console and  
> Voxeo?s logs ramp up when I dial the 2007 extension on X-lite but I  
> do not get any audio from the dialogue script in Voxeo?s Prophecy  
> ASR called ?Doctorsoffice.?
>
> Any ideas?
>
> Thanks.
>
> Mark.
>
> 0A
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mszlazak at aol.com  Wed Dec 10 10:55:21 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 13:55:21 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>
	<4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org>
Message-ID: <8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>


 Yes but someone else I'm in contact with set up FS a couple days ago and is having the same problems. 
Brian should I still update today?


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 10:45 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo









You're a month behind.. I highly recommend you update. ?Chances are this has already been fixed.





/b




On Dec 10, 2008, at 12:39 PM, mszlazak at aol.com wrote:




I?m trying to route calls from X-lite?<-->?FS (Nov. 6 2008?svn)?<-->?Voceo (Prophecy)?to use Voxeo?s ASR instead of FS?s built in PocketSphinx/ASR.

All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi=2 0router that connects a laptop to my desktop in case that matters but I?m not using the laptop for any of this.

??

I?ve set up an extension to bridge calls to Voxeo. Here is the entry?in file conf\dialplan\default.xml:

?

????<extension name="Doctors Office">

??????<condition field="destination_number" expression="^2007$">

????????<action application="info"/>

????????<action application="bridge" data="sofia/internal/sip:Doctorsoffice at 127.0.0.1:5068"/>???????

??????</condition>

????</extension>

?

I20hear one ring then a hang up. No errors in the FS console and Voxeo?s logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeo?s Prophecy ASR called ?Doctorsoffice.??





Any ideas?

?

Thanks.





Mark.


?
0A



Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse.?Start Listening Now!
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From brian at freeswitch.org  Wed Dec 10 11:02:49 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 13:02:49 -0600
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com>
	<4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org>
	<8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>
Message-ID: <74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org>

If you're not trying this on the latest code then yes I would update  
if possible.  Do you recall who you were talking too?

/b

On Dec 10, 2008, at 12:55 PM, mszlazak at aol.com wrote:

> Yes but someone else I'm in contact with set up FS a couple days ago  
> and is having the same problems.
> Brian should I still update today?
>

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From john at loopfx.com  Wed Dec 10 11:16:07 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 14:16:07 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>
	<87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>

I just emailed it to him.

Thanks!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Wednesday, December 10, 2008 1:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
wrote:
> I have a pcap, but I'm not able to see anything obviously wrong with
it.

We find that some equipment (in fact a lot of equipment) have features
that cause issues to be quite non-obvious, so perhaps you could give
the pcap to Brian for him to review. He's a total ace when it comes to
bug hunting.

-MC

>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian
> West
> Sent: Wednesday, December 10, 2008 12:59 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> would be most helpful to capture a pcap of the entire thing by itself
start
> to finish.
>
>
>
> /b
>
>
>
> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>
> No. I realize that's it's a B2BUA and that's exactly what we want.
>
>
>
> Everything with the transfer seems to work fine, except that there is
no
> audio.
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From mszlazak at aol.com  Wed Dec 10 11:51:30 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 14:51:30 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>
	<74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org>
Message-ID: <8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>


 It was someone from Voxeo support. I think John was the main person helping me with this.

I updated but things got worse all over.
I now can't run other extensions Gino's pizza or some db stuff

Got these errors:

2008-12-10 11:40:23 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll
**The specified module could not be found.? **
2008-12-10 11:40:23 [CONSOLE] mod_spidermonkey.c:944 sm_load_file() Successfully Loaded [C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_core_db.dll]
2008-12-10 11:40:23 [CONSOLE] mod_spidermonkey.c:944 sm_load_file() Successfully Loaded [C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_socket.dll]
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_spidermonkey]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'javascript'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'jsrun'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'jsapi'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_lua]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'lua'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'luarun'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'lua'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_say_en]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:371 switch_loadable_module_process() Adding Say interface 'en'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:118 switch_loadable_module_runtime() Starting runtime thread for CORE_SOFTTIMER_MODULE
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:118 switch_loadable_module_runtime() Starting runtime thread for mod_event_socket
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list dl-candidates default (allow)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 10.0.0.0/8 (deny) to list dl-candidates
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 172.16.0.0/12 (deny) to list dl-candidates
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.0.0/16 (deny) to list dl-candidates
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list rfc1918 default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 10.0.0.0/8 (allow) to list rfc1918
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 172.16.0.0/12 (allow) to list rfc1918
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.0.0/16 (allow) to list rfc1918
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list lan default (allow)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.42.0/24 (deny) to list lan
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.42.42/32 (allow) to list lan
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list strict default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 208.102.123.124/32 (allow) to list strict
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list domains default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:907 switch_load_network_lists() Adding 1.2.3.4/24 (allow) [brian at 10.0.0.2] to list domains
2008-12-10 11:40:23 [CONSOLE] switch_core.c:1258 switch_core_init_and_modload()
FreeSWITCH Version 1.0.trunk (10171M) Started.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]

freeswitch at mark-e40edd7b44> 2008-12-10 11:40:24 [ERR] sofia.c:543 sofia_profile_thread_run() Error Creating SIP UA for profile: internal-ipv6
2008-12-10 11:40:45 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/1000 at 10.0.0.2 [88d1aaf9-a625-444e-883d-c5ac6eeac30e]
2008-12-10 11:40:45 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch->2006 in context default
2008-12-10 11:40:45 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000 at 10.0.0.2] has been answered
2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
2008-12-10 11:40:48 [NOTICE] switch_ivr_async.c:1845 switch_ivr_detect_speech() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2008-12-10 11:40:48 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 1 (sofia/internal/1000 at 10.0.0.2) Ended
2008-12-10 11:40:48 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]
2008-12-10 11:41:23 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/1000 at 10.0.0.2 [67882be4-4c10-d248-8fe1-ac02b9b8fc5c]
2008-12-10 11:41:23 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch->8337 in context default
2008-12-10 11:41:23 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000 at 10.0.0.2] has been answered
2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!
2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine!
2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000 at 10.0.0.2) Ended
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]



 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 11:02 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo









If you're not trying this on the latest code then yes I would update if possible. ?Do you recall who you were talking too?



/b




On Dec 10, 2008, at 12:55 PM, mszlazak at aol.com wrote:



Yes but someone else I'm in contact with set up FS a couple days ago and is having the same problems.?
Brian should I still update today?






=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From brian at freeswitch.org  Wed Dec 10 11:59:02 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 13:59:02 -0600
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com>
	<74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org>
	<8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>
Message-ID: <D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org>

Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.

/b

On Dec 10, 2008, at 1:51 PM, mszlazak at aol.com wrote:

> It was someone from Voxeo support. I think John was the main person  
> helping me with this.
>
> I updated but things got worse all over.
> I now can't run other extensions Gino's pizza or some db stuff
>
>
> 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57  
> switch_core_asr_open() Invalid ASR module [pocketsphinx]!
>  2008-12-10 11:41:23 [ERR] switch_core_speech.c:60  
> switch_core_speech_open() Invalid speech module [openmrcp]!
> 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859  
> init_speech_engine() Invalid TTS module!
> 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()  Cannot  
> allocate speech engine!
> 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160  
> switch_core_standard_on_execute() Hangup sofia/internal/ 
> 1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
> 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878  
> switch_core_session_thread() Session 2 (sofia/internal/ 
> 1000 at 10.0.0.2) Ended
> 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880  
> switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 
>  [CS_HANGUP]
>
>
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From brian at freeswitch.org  Wed Dec 10 12:27:22 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 14:27:22 -0600
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
Message-ID: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>

FreeSWITCHers,

I'm looking for donations for the next batch of sound files we need to  
have done for the up coming 1.0.2 release.

I have had others pitch in some money in the past and I thank everyone  
for doing so.  I hope everyone
can come together and help me raise about $200 to pay for this batch  
of prompts.

I also would like to thank Bandwidth.com and Teliax for their support  
of the FreeSWITCH project.

Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...

If you wish to donate please paypal brian at freeswitch.org that'll help  
out!

Happy Holidays,
Brian West
FreeSWITCH.org
PS: If you know of any sound files we need let me know.



From mszlazak at aol.com  Wed Dec 10 12:37:25 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 15:37:25 -0500
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
In-Reply-To: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
Message-ID: <8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>


 
How do I donate?


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Cc: freeswitch-dev at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 12:27 pm
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware










FreeSWITCHers,

I'm looking for donations for the next batch of sound files we need to  
have done for the up coming 1.0.2 release.

I have had others pitch in some money in the past and I thank everyone  
for doing so.  I hope everyone
can come together and help me raise about $200 to pay for this batch  
of prompts.

I also would like to thank Bandwidth.com and Teliax for their support  
of the FreeSWITCH project.

Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...

If you wish to donate please paypal brian at freeswitch.org that'll help  
out!

Happy Holidays,
Brian West
FreeSWITCH.org
PS: If you know of any sound files we need let me know.

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From intralanman at freeswitch.org  Wed Dec 10 12:41:27 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 10 Dec 2008 15:41:27 -0500
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>	<87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>
Message-ID: <49402977.4090808@freeswitch.org>

try blocking ICMP packets TO the MSS.... i had this exact same problem a 
few months ago.... MSS starts sending RTP to FS before FS is ready to 
accept.... so the OS catches the port not open and returns an ICMP 3:3 
back to the MSS.... which in turn chokes on the queued up RTP and 
refuses to send anymore...

-Ray

John Rutherford wrote:
> I just emailed it to him.
>
> Thanks!
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
> Sent: Wednesday, December 10, 2008 1:09 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
> On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
> wrote:
>   
>> I have a pcap, but I'm not able to see anything obviously wrong with
>>     
> it.
>
> We find that some equipment (in fact a lot of equipment) have features
> that cause issues to be quite non-obvious, so perhaps you could give
> the pcap to Brian for him to review. He's a total ace when it comes to
> bug hunting.
>
> -MC
>
>   
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>     
> Brian
>   
>> West
>> Sent: Wednesday, December 10, 2008 12:59 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] No audio after transfer
>>
>>
>>
>> would be most helpful to capture a pcap of the entire thing by itself
>>     
> start
>   
>> to finish.
>>
>>
>>
>> /b
>>
>>
>>
>> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>>
>> No. I realize that's it's a B2BUA and that's exactly what we want.
>>
>>
>>
>> Everything with the transfer seems to work fine, except that there is
>>     
> no
>   
>> audio.
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From brian at freeswitch.org  Wed Dec 10 12:43:15 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 14:43:15 -0600
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
In-Reply-To: <8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
Message-ID: <4B1AFF7A-F3DF-41B6-ADF3-14767B44BA41@freeswitch.org>

Paypal works great!  ;)

/b

On Dec 10, 2008, at 2:37 PM, mszlazak at aol.com wrote:

> If you wish to donate please paypal brian at freeswitch.org that'll help

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From intralanman at freeswitch.org  Wed Dec 10 12:45:25 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 10 Dec 2008 15:45:25 -0500
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
In-Reply-To: <8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
Message-ID: <49402A65.7090808@freeswitch.org>

i saw this in the first email

"If you wish to donate please paypal brian at freeswitch.org 
<mailto:brian at freeswitch.org> that'll help "

-Ray


mszlazak at aol.com wrote:
>
> How do I donate?
>
> -----Original Message-----
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Cc: freeswitch-dev at lists.freeswitch.org
> Sent: Wed, 10 Dec 2008 12:27 pm
> Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
>
> FreeSWITCHers,
>
>
>
> I'm looking for donations for the next batch of sound files we need to  
>
> have done for the up coming 1.0.2 release.
>
>
>
> I have had others pitch in some money in the past and I thank everyone  
>
> for doing so.  I hope everyone
>
> can come together and help me raise about $200 to pay for this batch  
>
> of prompts.
>
>
>
> I also would like to thank Bandwidth.com and Teliax for their support  
>
> of the FreeSWITCH project.
>
>
>
> Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...
>
>
>
> If you wish to donate please paypal brian at freeswitch.org <mailto:brian at freeswitch.org> that'll help  
>
> out!
>
>
>
> Happy Holidays,
>
> Brian West
>
> FreeSWITCH.org
>
> PS: If you know of any sound files we need let me know.
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org <mailto:Freeswitch-users at lists.freeswitch.org>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
> Listen to 350+ music, sports, & news radio stations -- including songs 
> for the holidays -- FREE while you browse. Start Listening Now 
> <http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mszlazak at aol.com  Wed Dec 10 13:24:31 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 16:24:31 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com><74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org><8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>
	<D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org>
Message-ID: <8CB2924D2F19E7B-9D4-9EE@Webmail-mg06.sim.aol.com>


 Yup my bad. 

But I'm still getting this error:

2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll
**The specified module could not be found.? **

It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then?

I'm still having the same problem with no audio from Voxeo.

Mark. 


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 11:59 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo









Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.



/b




On Dec 10, 2008, at 1:51 PM, mszlazak at aol.com wrote:



 It was someone from Voxeo support. I think John was the main person helping me with this.
 
 I updated but things got worse all over.
 I now can't run other extensions Gino's pizza or some db stuff
 

 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
?2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!
 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine!
 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000 at 10.0.0.2) Ended
 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]
 
 
 





=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From jaugenstine at gmail.com  Wed Dec 10 13:33:01 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Wed, 10 Dec 2008 13:33:01 -0800
Subject: [Freeswitch-users]  conference module question - prompts
Message-ID: <207e7a5e0812101333k28b27e52tbfe852458eae204f@mail.gmail.com>

I am trying to modify the behavior of the playing of prompts when someone
enters the conference.  When I enable the conf-welcome prompt and a new
participant enters the conference, the prompt is played to the conference
and everyone hears the welcome.  Is there a any way to modify the
configuration so that the welcome prompt is only heard by the participant
entering the conference?

Thank you.
Jonathan
jaugenstine at gmail.com
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From john at loopfx.com  Wed Dec 10 14:18:52 2008
From: john at loopfx.com (John Rutherford)
Date: Wed, 10 Dec 2008 17:18:52 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>	<87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>
	<49402977.4090808@freeswitch.org>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>

Okay. I just tried this.  

 

Now we're getting the audio going one way, but not the other.  So, I can
hear the person that I just transferred to, but they can't hear me.

 

Anyone have any other ideas?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Raymond Chandler
Sent: Wednesday, December 10, 2008 3:41 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

try blocking ICMP packets TO the MSS.... i had this exact same problem a
few months ago.... MSS starts sending RTP to FS before FS is ready to
accept.... so the OS catches the port not open and returns an ICMP 3:3
back to the MSS.... which in turn chokes on the queued up RTP and
refuses to send anymore...

-Ray

John Rutherford wrote: 

I just emailed it to him.
 
Thanks!
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Wednesday, December 10, 2008 1:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer
 
On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
<mailto:john at loopfx.com> 
wrote:
  

	I have a pcap, but I'm not able to see anything obviously wrong
with
	    

it.
 
We find that some equipment (in fact a lot of equipment) have features
that cause issues to be quite non-obvious, so perhaps you could give
the pcap to Brian for him to review. He's a total ace when it comes to
bug hunting.
 
-MC
 
  

	 
	 
	From: freeswitch-users-bounces at lists.freeswitch.org
	[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
	    

Brian
  

	West
	Sent: Wednesday, December 10, 2008 12:59 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] No audio after transfer
	 
	 
	 
	would be most helpful to capture a pcap of the entire thing by
itself
	    

start
  

	to finish.
	 
	 
	 
	/b
	 
	 
	 
	On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
	 
	No. I realize that's it's a B2BUA and that's exactly what we
want.
	 
	 
	 
	Everything with the transfer seems to work fine, except that
there is
	    

no
  

	audio.
	 
	 
	 
	_______________________________________________
	Freeswitch-users mailing list
	Freeswitch-users at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	http://www.freeswitch.org
	 
	 
	    

 
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

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From carlos.talbot at gmail.com  Wed Dec 10 14:57:43 2008
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Wed, 10 Dec 2008 16:57:43 -0600
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <8CB2924D2F19E7B-9D4-9EE@Webmail-mg06.sim.aol.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com><74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org><8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com>
	<D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org>
	<8CB2924D2F19E7B-9D4-9EE@Webmail-mg06.sim.aol.com>
Message-ID: <587BE2B8-E458-4050-B3AA-46341A2B52B1@gmail.com>

There was a typecast warning that prevented spidermoneky from  
compiling in a recent svn. Did you check to see if it compiled?

Sent from my iPhone

On Dec 10, 2008, at 3:24 PM, mszlazak at aol.com wrote:

> Yup my bad.
>
> But I'm still getting this error:
>
> 2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file()  
> Error Loading module C:\Source\freeswitch-snapshot\Debug\mod 
> \mod_spidermonkey_teletone.dll
> **The specified module could not be found.  **
>
> It doesn't look like it was put in this latest snapshot. I could use  
> that dll from my older snapshot, has it been changed since then?
>
> I'm still having the same problem with no audio from Voxeo.
>
> Mark.
>
>
> -----Original Message-----
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wed, 10 Dec 2008 11:59 am
> Subject: Re: [Freeswitch-users] Audio routing problem between FS and  
> Voxeo
>
> Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.
>
> /b
>
> On Dec 10, 2008, at 1:51 PM, mszlazak at aol.com wrote:
>
>> It was someone from Voxeo support. I think John was the main person  
>> helping me with this.
>>
>> I updated but things got worse all over.
>> I now can't run other extensions Gino's pizza or some db stuff
>>
>>
>> 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57  
>> switch_core_asr_open() Invalid ASR module [pocketsphinx]!
>>  2008-12-10 11:41:23 [ERR] switch_core_speech.c:60  
>> switch_core_speech_open() Invalid speech module [openmrcp]!
>> 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859  
>> init_speech_engine() Invalid TTS module!
>> 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()  Cannot  
>> allocate speech engine!
>> 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160  
>> switch_core_standard_on_execute() Hangup sofia/internal/ 
>> 1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878  
>> switch_core_session_thread() Session 2 (sofia/internal/ 
>> 1000 at 10.0.0.2) Ended
>> 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880  
>> switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 
>>  [CS_HANGUP]
>>
>>
> =
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> Listen to 350+ music, sports, & news radio stations ? including song 
> s for the holidays ? FREE while you browse. Start Listening Now!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Wed Dec 10 14:52:52 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 14:52:52 -0800
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local>
	<191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local>
	<BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org>
	<81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local>
	<87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local>
	<49402977.4090808@freeswitch.org>
	<81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>
Message-ID: <87f2f3b90812101452t7076bb6fo93f7a78bbfb0404f@mail.gmail.com>

I smell a NAT... is there any NAT involved?

On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford <john at loopfx.com> wrote:
> Okay. I just tried this.
>
>
>
> Now we're getting the audio going one way, but not the other.  So, I can
> hear the person that I just transferred to, but they can't hear me.
>
>
>
> Anyone have any other ideas?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raymond
> Chandler
> Sent: Wednesday, December 10, 2008 3:41 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> try blocking ICMP packets TO the MSS.... i had this exact same problem a few
> months ago.... MSS starts sending RTP to FS before FS is ready to accept....
> so the OS catches the port not open and returns an ICMP 3:3 back to the
> MSS.... which in turn chokes on the queued up RTP and refuses to send
> anymore...
>
> -Ray
>
> John Rutherford wrote:
>
> I just emailed it to him.
>
>
>
> Thanks!
>
>
>
> -----Original Message-----
>
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>
> Michael Collins
>
> Sent: Wednesday, December 10, 2008 1:09 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
>
> wrote:
>
>
>
> I have a pcap, but I'm not able to see anything obviously wrong with
>
>
>
> it.
>
>
>
> We find that some equipment (in fact a lot of equipment) have features
>
> that cause issues to be quite non-obvious, so perhaps you could give
>
> the pcap to Brian for him to review. He's a total ace when it comes to
>
> bug hunting.
>
>
>
> -MC
>
>
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>
>
>
> Brian
>
>
>
> West
>
> Sent: Wednesday, December 10, 2008 12:59 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
>
>
>
>
> would be most helpful to capture a pcap of the entire thing by itself
>
>
>
> start
>
>
>
> to finish.
>
>
>
>
>
>
>
> /b
>
>
>
>
>
>
>
> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>
>
>
> No. I realize that's it's a B2BUA and that's exactly what we want.
>
>
>
>
>
>
>
> Everything with the transfer seems to work fine, except that there is
>
>
>
> no
>
>
>
> audio.
>
>
>
>
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mszlazak at aol.com  Wed Dec 10 15:01:11 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 18:01:11 -0500
Subject: [Freeswitch-users]  Audio routing problem between FS and Voxeo
Message-ID: <8CB2932540232CF-9D4-1024@Webmail-mg06.sim.aol.com>

Brian, I'm still having the same audio problem when bridging/routing to Voxeo using the latest snapshot. Help!





Also, it doesn't look like mod_spidermonkey_teletone.dll was
put into this latest snapshot. Has it been changed since November? I could use that older version of this dll.











Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 





 


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From brian at freeswitch.org  Wed Dec 10 15:06:17 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 17:06:17 -0600
Subject: [Freeswitch-users] Hardware requests.
Message-ID: <1B94AB6C-022D-47E8-9868-92CE89EA0FD4@freeswitch.org>

Does anyone have a Polycom 320 they would like to donate to the  
project?  I need one to trouble shoot a problem.  Please contact me  
off list.

Thanks,
Brian West




From mszlazak at aol.com  Wed Dec 10 15:48:01 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 18:48:01 -0500
Subject: [Freeswitch-users] Audio routing problem between FS and Voxeo
In-Reply-To: <587BE2B8-E458-4050-B3AA-46341A2B52B1@gmail.com>
References: <8CB290DCD02A172-1704-ADC@MBLK-M37.sysops.aol.com><4E0581E7-ED84-4525-80B4-6C7D4991FDFD@freeswitch.org><8CB290FFC88D892-1704-C02@MBLK-M37.sysops.aol.com><74D348CB-C4AE-44CC-A331-3727B6927BD2@freeswitch.org><8CB2917D4EB647B-9D4-347@Webmail-mg06.sim.aol.com><D626A66E-DBD0-43F0-A5A7-D2A0FF9751A4@freeswitch.org><8CB2924D2F19E7B-9D4-9EE@Webmail-mg06.sim.aol.com>
	<587BE2B8-E458-4050-B3AA-46341A2B52B1@gmail.com>
Message-ID: <8CB2938DF2CC0FD-9D4-12BD@Webmail-mg06.sim.aol.com>

I only glanced at the
compilers output when it ended and it reported no no errors but I did
not look at the warnings along the way. Have things been modified since
so spider monkey compiles?? 



More importantly, do you have any ideas as what is going on with my audio problem. I can't attach a wireshark .pcap file since it's to big for your list and my email gets rejected.

 


 

-----Original Message-----
From: Carlos Talbot <carlos.talbot at gmail.com>
To: freeswitch-users at lists.freeswitch.org <freeswitch-users at lists.freeswitch.org>
Cc: freeswitch-users at lists.freeswitch.org <freeswitch-users at lists.freeswitch.org>
Sent: Wed, 10 Dec 2008 2:57 pm
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo










There was a typecast warning that prevented spidermoneky from compiling in a recent svn. Did you check to see if it compiled?

Sent from my iPhone


On Dec 10, 2008, at 3:24 PM, mszlazak at aol.com wrote:








 Yup my bad. 



But I'm still getting this error:



2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll

**The specified module could not be found.? **



It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then?



I'm still having the same problem with no audio from Voxeo.



Mark. 





 





 



-----Original Message-----

From: Brian West <brian at frees
witch.org>

To: freeswitch-users at lists.freeswitch.org

Sent: Wed, 10 Dec 2008 11:59 am

Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo












Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.







/b









On Dec 10, 2008, at 1:51 PM, mszlazak at aol.com wrote:





 It was someone from Voxeo support. I think John was the main person helping me with this.

 

 I updated but things got worse all over.

 I now can't run other extensions Gino's pizza or some db stuff

 



 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!

?2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!

 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!

 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine!

 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000 at 10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]

 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000 at 10.0.0.2) Ended

 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000 at 10.0.0.2 [CS_HANGUP]

 

 

 










=



 






_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitc
h-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





 





Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From anthony.minessale at gmail.com  Wed Dec 10 16:48:19 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Dec 2008 18:48:19 -0600
Subject: [Freeswitch-users] conference module question - prompts
In-Reply-To: <207e7a5e0812101333k28b27e52tbfe852458eae204f@mail.gmail.com>
References: <207e7a5e0812101333k28b27e52tbfe852458eae204f@mail.gmail.com>
Message-ID: <191c3a030812101648k53435801iaa3e4a3ba26560e5@mail.gmail.com>

not currently,  no

On Wed, Dec 10, 2008 at 3:33 PM, jonathan augenstine
<jaugenstine at gmail.com>wrote:

> I am trying to modify the behavior of the playing of prompts when someone
> enters the conference.  When I enable the conf-welcome prompt and a new
> participant enters the conference, the prompt is played to the conference
> and everyone hears the welcome.  Is there a any way to modify the
> configuration so that the welcome prompt is only heard by the participant
> entering the conference?
>
> Thank you.
> Jonathan
> jaugenstine at gmail.com
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From krice at suspicious.org  Wed Dec 10 17:39:32 2008
From: krice at suspicious.org (Ken Rice)
Date: Wed, 10 Dec 2008 19:39:32 -0600
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for pending
	1.0.2/Hardware
In-Reply-To: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
Message-ID: <C565CB74.4AFF2%krice@suspicious.org>

Hey Guys... I spoke with Brian on this a few minutes ago and some money has
already showed up for the sound files... Lets see if we can go ahead and
make it where Brian can get these files on order early tomorrow so get can
make sure they get us a good Christmas Present in the form of a new stable
release

K 


> From: Brian West <brian at freeswitch.org>
> Reply-To: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Date: Wed, 10 Dec 2008 14:27:22 -0600
> To: <freeswitch-users at lists.freeswitch.org>
> Cc: "freeswitch-dev at lists.freeswitch.org"
> <freeswitch-dev at lists.freeswitch.org>
> Subject: [Freeswitch-dev] Sounds for pending 1.0.2/Hardware
> 
> FreeSWITCHers,
> 
> I'm looking for donations for the next batch of sound files we need to
> have done for the up coming 1.0.2 release.
> 
> I have had others pitch in some money in the past and I thank everyone
> for doing so.  I hope everyone
> can come together and help me raise about $200 to pay for this batch
> of prompts.
> 
> I also would like to thank Bandwidth.com and Teliax for their support
> of the FreeSWITCH project.
> 
> Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...
> 
> If you wish to donate please paypal brian at freeswitch.org that'll help
> out!
> 
> Happy Holidays,
> Brian West
> FreeSWITCH.org
> PS: If you know of any sound files we need let me know.
> 
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org





From ack at telefonica.net  Wed Dec 10 17:51:56 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Thu, 11 Dec 2008 02:51:56 +0100
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for
	pending	1.0.2/Hardware
In-Reply-To: <C565CB74.4AFF2%krice@suspicious.org>
References: <C565CB74.4AFF2%krice@suspicious.org>
Message-ID: <1228960316.10071.21.camel@develop4>

I'm in too . Brian hope you got money i sent, a pleasure to contribute.

Cheers, ( sack )

El mi?, 10-12-2008 a las 19:39 -0600, Ken Rice escribi?:
> Hey Guys... I spoke with Brian on this a few minutes ago and some money has
> already showed up for the sound files... Lets see if we can go ahead and
> make it where Brian can get these files on order early tomorrow so get can
> make sure they get us a good Christmas Present in the form of a new stable
> release
> 
> K 
> 
> 
> > From: Brian West <brian at freeswitch.org>
> > Reply-To: "freeswitch-dev at lists.freeswitch.org"
> > <freeswitch-dev at lists.freeswitch.org>
> > Date: Wed, 10 Dec 2008 14:27:22 -0600
> > To: <freeswitch-users at lists.freeswitch.org>
> > Cc: "freeswitch-dev at lists.freeswitch.org"
> > <freeswitch-dev at lists.freeswitch.org>
> > Subject: [Freeswitch-dev] Sounds for pending 1.0.2/Hardware
> > 
> > FreeSWITCHers,
> > 
> > I'm looking for donations for the next batch of sound files we need to
> > have done for the up coming 1.0.2 release.
> > 
> > I have had others pitch in some money in the past and I thank everyone
> > for doing so.  I hope everyone
> > can come together and help me raise about $200 to pay for this batch
> > of prompts.
> > 
> > I also would like to thank Bandwidth.com and Teliax for their support
> > of the FreeSWITCH project.
> > 
> > Are you ready for 1.0.2?  Go download SVN Trunk and beat it up for us...
> > 
> > If you wish to donate please paypal brian at freeswitch.org that'll help
> > out!
> > 
> > Happy Holidays,
> > Brian West
> > FreeSWITCH.org
> > PS: If you know of any sound files we need let me know.
> > 
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

"No basta saber, hay que aplicar lo que se sabe; 
no basta querer hacerlas cosas, hay que hacerlas".

"Knowing is not enough; we must apply. 
 Willing is not enough; we must do"

 Johann Wolfgang von Goethe
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From brian at freeswitch.org  Wed Dec 10 17:56:33 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2008 19:56:33 -0600
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for
	pending	1.0.2/Hardware
In-Reply-To: <1228960316.10071.21.camel@develop4>
References: <C565CB74.4AFF2%krice@suspicious.org>
	<1228960316.10071.21.camel@develop4>
Message-ID: <C114E867-AE3B-411F-912B-A3B203C4AF8A@freeswitch.org>

Thank you it really helps.  I want to make sure the 1.0.2 release is  
the best release ever!

/b

On Dec 10, 2008, at 7:51 PM, Angel Carpintero wrote:

> I'm in too . Brian hope you got money i sent, a pleasure to  
> contribute.




From mszlazak at aol.com  Wed Dec 10 18:23:57 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 10 Dec 2008 21:23:57 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for pending
	1.0.2/Hardware
In-Reply-To: <C114E867-AE3B-411F-912B-A3B203C4AF8A@freeswitch.org>
References: <C565CB74.4AFF2%krice@suspicious.org><1228960316.10071.21.camel@develop4>
	<C114E867-AE3B-411F-912B-A3B203C4AF8A@freeswitch.org>
Message-ID: <8CB294EA800ADED-9D4-1AE7@Webmail-mg06.sim.aol.com>


 Good enough will do to get my cash. 
It will be on the way once my paypal account is confirmed in a few days.


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 10 Dec 2008 5:56 pm
Subject: Re: [Freeswitch-users] [Freeswitch-dev] Sounds for pending 1.0.2/Hardware










Thank you it really helps.  I want to make sure the 1.0.2 release is  
the best release ever!

/b

On Dec 10, 2008, at 7:51 PM, Angel Carpintero wrote:

> I'm in too . Brian hope you got money i sent, a pleasure to  
> contribute.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From edpimentl at gmail.com  Wed Dec 10 19:49:55 2008
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 10 Dec 2008 22:49:55 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for pending
	1.0.2/Hardware
In-Reply-To: <C565CB74.4AFF2%krice@suspicious.org>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<C565CB74.4AFF2%krice@suspicious.org>
Message-ID: <9dc4a1670812101949m7864aeb9m1bb8b069cff8fadf@mail.gmail.com>

Bryan!!,

Count on me too.
E
http://Gpro.ws
http://DatR.ws (Store, Sync, Share, Publish)
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From edpimentl at gmail.com  Wed Dec 10 20:41:22 2008
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 10 Dec 2008 23:41:22 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Sounds for pending
	1.0.2/Hardware
In-Reply-To: <C565CB74.4AFF2%krice@suspicious.org>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<C565CB74.4AFF2%krice@suspicious.org>
Message-ID: <9dc4a1670812102041p4c5e2af8i77ae5d8ef4a9f1c@mail.gmail.com>

Here is some "sound advice" regarding some unique sounds links:

http://flashkit.com/soundfx/
Flash Kit has an ever growing list of shareware and freeware SoundFX for
download. It provides a powerful searching, excellent organisation and easy
real time  previews make this the most advanced Sound FX download site on
the net!  mp3 and flashtrak versions of all effects as well!

http://www.audiomicro.com/royalty-free-sound-effects.html
AudioMicro is a revolutionary collection of user-generated micro stock
music, sound effects, production music, production elements and music  cues.
Finally, high quality audio content is available at unbelievable  prices
with no hidden costs or fees.

http://www.sound-effect.com/
Search for, preview and download royalty free sound effects for immediate
use in your multimedia project. These high quality royalty free sound
effects are hand-picked from only the best sound designers

http://www.soundsnap.com/browse
Soundsnap is the best platform to find and share free sound effects and
loops- legally. It is a collection of original sounds made or recorded  by
its users, and not songs or sound FX found on commercial libraries  or
sample CD


http://www.findsounds.com/
FindSounds.com is a free site for finding sound effects and musical
instrument samples on the Web. It is a Web search engine, like Google and
AltaVista, but with a focus on sounds. It provides powerful features, yet is
simple and easy to use, and suitable for all ages

http://soungle.com/
Soungle is a free site, developed by Southern Codes, for finding all  kind
of sound FX and musical instruments samples on our mega online  library. As
different from most of similar sites, Soungle is NOT a Web  search engine.
It only searches in our growing monster database. Our  goals are to keep it
simple to use (search, preview and download) and  to keep it free

http://www.sfxsource.com/Sound-Effects
In SFX Source youll find cutting-edge and imaginative sound samples crafted
with  passionate expertise for use in all levels of production, from
professional to amateur, for use in Film, TV, Games, and New Media.


http://www.a1freesoundeffects.com/
A1 Free Sound Effects wants to provide the internet with our Free Sound
Effects that you can download to your computer and use for church, school,
home or for any non-profit project. Commercial Sounds Available.


http://soundrangers.com/html/free-sound-effects.html
Soundrangers specializes in generating state-of-the-art royalty free  sound
effects and music for interactive media, such as virtual  user-interfaces,
games, online entertainment, web sites and  communication devices.

http://www.partnersinrhyme.com/pir/PIRsfx.shtml
Partners In Rhyme provide free  resources, help and advice to amateur and
professional multimedia  producers, film makers, musicians and students in
their search for  music, sound effects and audio tools to complete their
projects  inexpensively and quickly

http://www.soundboard.com/
Soundboard.com puts all your audio clips, as well as millions from other
users in one, easy to manage location.  Its mission is to invite everyone to
help us create a central site for audio clips in a format that anyone with a
connection and a browser can enjoy

E
http://Gpro.ws
http://DatR.ws (Store, Sync, Share, Publish)
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From carole.olivier at enst.fr  Wed Dec 10 22:51:41 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 10 Dec 2008 22:51:41 -0800 (PST)
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
Message-ID: <20950299.post@talk.nabble.com>


Hello,

I am running the version 1.0.1.
Do you still need me to run the debugging?

Carole


Brian West-3 wrote:
> 
> I have already labbed this up on SVN trunk and I don't get a segfault  
> but I get something else that prevents it from working properly.    We  
> are working on it today.  Also what version are you running?
> 
> /b
> 
> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
> 
>> Thanks for reporting this. It would be helpful to know a bit more. Can
>> you start freeswitch and press F12 (or type "version" at the CLI) and
>> report back what it says?
>> Also, a backtrace (bt) is generally useful. If you could produce a
>> "bt" and a "bt full" from you core file that would be extremely
>> helpful.
>>
>> see this link for more information:
>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>
>> you should have a "core" file for each segfault that occurred. Use the
>> gdb program to get the back trace:
>>
>> gdb /path/to/fs/binary core.xxx
>>
>> then capture the output from these two commands:
>>
>> bt <enter>
>> bt full <enter>
>>
>> When you type those commands you'll see tons of debugging info;
>> capture that and put it in a pastebin (pastebin.freeswitch.org) then
>> report back here.
>>
>> You can exit the gdb debugger by typing q <enter>
>>
>> Thanks for helping us collect information!
>>
>> -MC
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From msc at freeswitch.org  Wed Dec 10 23:05:16 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Dec 2008 23:05:16 -0800
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20950299.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
Message-ID: <87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>

Carole,

There have been many updates since 1.0.1 was officially released. If
you could start FreeSWITCH and then press F12 it will reveal which SVN
revisionnumber you are running. Please supply that number and it will
help us to know if you are on a recent revision.

Thanks,
MC

On Wed, Dec 10, 2008 at 10:51 PM, Carole O. <carole.olivier at enst.fr> wrote:
>
> Hello,
>
> I am running the version 1.0.1.
> Do you still need me to run the debugging?
>
> Carole
>
>
> Brian West-3 wrote:
>>
>> I have already labbed this up on SVN trunk and I don't get a segfault
>> but I get something else that prevents it from working properly.    We
>> are working on it today.  Also what version are you running?
>>
>> /b
>>
>> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
>>
>>> Thanks for reporting this. It would be helpful to know a bit more. Can
>>> you start freeswitch and press F12 (or type "version" at the CLI) and
>>> report back what it says?
>>> Also, a backtrace (bt) is generally useful. If you could produce a
>>> "bt" and a "bt full" from you core file that would be extremely
>>> helpful.
>>>
>>> see this link for more information:
>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>
>>> you should have a "core" file for each segfault that occurred. Use the
>>> gdb program to get the back trace:
>>>
>>> gdb /path/to/fs/binary core.xxx
>>>
>>> then capture the output from these two commands:
>>>
>>> bt <enter>
>>> bt full <enter>
>>>
>>> When you type those commands you'll see tons of debugging info;
>>> capture that and put it in a pastebin (pastebin.freeswitch.org) then
>>> report back here.
>>>
>>> You can exit the gdb debugger by typing q <enter>
>>>
>>> Thanks for helping us collect information!
>>>
>>> -MC
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> --
> View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From carole.olivier at enst.fr  Wed Dec 10 23:33:29 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 10 Dec 2008 23:33:29 -0800 (PST)
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
	<87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>
Message-ID: <20950711.post@talk.nabble.com>


I have got the following:
Freeswitch Version 1.0.1 (9171)

Carole



Michael Collins-11 wrote:
> 
> Carole,
> 
> There have been many updates since 1.0.1 was officially released. If
> you could start FreeSWITCH and then press F12 it will reveal which SVN
> revisionnumber you are running. Please supply that number and it will
> help us to know if you are on a recent revision.
> 
> Thanks,
> MC
> 
> On Wed, Dec 10, 2008 at 10:51 PM, Carole O. <carole.olivier at enst.fr>
> wrote:
>>
>> Hello,
>>
>> I am running the version 1.0.1.
>> Do you still need me to run the debugging?
>>
>> Carole
>>
>>
>> Brian West-3 wrote:
>>>
>>> I have already labbed this up on SVN trunk and I don't get a segfault
>>> but I get something else that prevents it from working properly.    We
>>> are working on it today.  Also what version are you running?
>>>
>>> /b
>>>
>>> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
>>>
>>>> Thanks for reporting this. It would be helpful to know a bit more. Can
>>>> you start freeswitch and press F12 (or type "version" at the CLI) and
>>>> report back what it says?
>>>> Also, a backtrace (bt) is generally useful. If you could produce a
>>>> "bt" and a "bt full" from you core file that would be extremely
>>>> helpful.
>>>>
>>>> see this link for more information:
>>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>>
>>>> you should have a "core" file for each segfault that occurred. Use the
>>>> gdb program to get the back trace:
>>>>
>>>> gdb /path/to/fs/binary core.xxx
>>>>
>>>> then capture the output from these two commands:
>>>>
>>>> bt <enter>
>>>> bt full <enter>
>>>>
>>>> When you type those commands you'll see tons of debugging info;
>>>> capture that and put it in a pastebin (pastebin.freeswitch.org) then
>>>> report back here.
>>>>
>>>> You can exit the gdb debugger by typing q <enter>
>>>>
>>>> Thanks for helping us collect information!
>>>>
>>>> -MC
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> --
>> View this message in context:
>> http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950711.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From msc at freeswitch.org  Wed Dec 10 23:58:44 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 10 Dec 2008 23:58:44 -0800
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20950711.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
	<87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>
	<20950711.post@talk.nabble.com>
Message-ID: <7DF34003-ACE0-496A-B986-E91E83C75C4B@freeswitch.org>

Wow! That is really old. I strongly recommend that you update to the  
latest trunk using svn.

-MC


On Dec 10, 2008, at 11:33 PM, "Carole O." <carole.olivier at enst.fr>  
wrote:

>
> I have got the following:
> Freeswitch Version 1.0.1 (9171)
>
> Carole
>
>
>
> Michael Collins-11 wrote:
>>
>> Carole,
>>
>> There have been many updates since 1.0.1 was officially released. If
>> you could start FreeSWITCH and then press F12 it will reveal which  
>> SVN
>> revisionnumber you are running. Please supply that number and it will
>> help us to know if you are on a recent revision.
>>
>> Thanks,
>> MC
>>
>> On Wed, Dec 10, 2008 at 10:51 PM, Carole O. <carole.olivier at enst.fr>
>> wrote:
>>>
>>> Hello,
>>>
>>> I am running the version 1.0.1.
>>> Do you still need me to run the debugging?
>>>
>>> Carole
>>>
>>>
>>> Brian West-3 wrote:
>>>>
>>>> I have already labbed this up on SVN trunk and I don't get a  
>>>> segfault
>>>> but I get something else that prevents it from working  
>>>> properly.    We
>>>> are working on it today.  Also what version are you running?
>>>>
>>>> /b
>>>>
>>>> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
>>>>
>>>>> Thanks for reporting this. It would be helpful to know a bit  
>>>>> more. Can
>>>>> you start freeswitch and press F12 (or type "version" at the  
>>>>> CLI) and
>>>>> report back what it says?
>>>>> Also, a backtrace (bt) is generally useful. If you could produce a
>>>>> "bt" and a "bt full" from you core file that would be extremely
>>>>> helpful.
>>>>>
>>>>> see this link for more information:
>>>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>>>
>>>>> you should have a "core" file for each segfault that occurred.  
>>>>> Use the
>>>>> gdb program to get the back trace:
>>>>>
>>>>> gdb /path/to/fs/binary core.xxx
>>>>>
>>>>> then capture the output from these two commands:
>>>>>
>>>>> bt <enter>
>>>>> bt full <enter>
>>>>>
>>>>> When you type those commands you'll see tons of debugging info;
>>>>> capture that and put it in a pastebin (pastebin.freeswitch.org)  
>>>>> then
>>>>> report back here.
>>>>>
>>>>> You can exit the gdb debugger by typing q <enter>
>>>>>
>>>>> Thanks for helping us collect information!
>>>>>
>>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950711.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From yudha2008 at gmail.com  Thu Dec 11 00:51:09 2008
From: yudha2008 at gmail.com (Baskar)
Date: Thu, 11 Dec 2008 14:21:09 +0530
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <191c3a030812030924u457f934ep77bd70680f583fcd@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
	<d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
	<191c3a030812030924u457f934ep77bd70680f583fcd@mail.gmail.com>
Message-ID: <d0640d380812110051w41bc5481r60a3d3242044d3f7@mail.gmail.com>

*Hi,

    when in dial from console inbound is working fine when i dial outbound
it is not working in console dialing.
*

FreeSWITCH Version 1.0.trunk (10567) Started.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
2008-12-11 14:17:03 [CONSOLE] mod_local_stream.c:142 read_stream_thread()
Can't open directory: /usr/local/freeswitch/sounds/music/16000

*freeswitch at localhost> pa devlist*
API CALL [pa(devlist)] output:
0;/dev/dsp;16;4
1;Intel ICH5: Intel ICH5 (hw:0,0);2;6
2;Intel ICH5: Intel ICH5 - MIC ADC (hw:0,1);2;0
3;Intel ICH5: Intel ICH5 - MIC2 ADC (hw:0,2);2;0
4;Intel ICH5: Intel ICH5 - ADC2 (hw:0,3);2;0
5;Intel ICH5: Intel ICH5 - IEC958 (hw:0,4);0;2
6;front;0;6
7;surround40;0;4
8;surround41;0;128
9;surround50;0;128
10;surround51;0;6
11;iec958;0;2
12;spdif;0;2
13;default;128;128
14;dmix;0;2
*
After that i dial 3 in softphone
Output:*

freeswitch at localhost> 2008-12-11 14:17:16 [NOTICE] switch_channel.c:564
switch_channel_set_name() New Channel
sofia/internal/1003 at 172.20.177.117[d49468e5-be90-40f6-8ffe-58c56651d87a]
2008-12-11 14:17:16 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
FreeSwitch->3 in context default
2008-12-11 14:17:16 [NOTICE] switch_channel.c:564 switch_channel_set_name()
New Channel portaudio/1003 [bfc0023c-0725-40a8-a187-574bdab40c40]
2008-12-11 14:17:17 [NOTICE] mod_portaudio.c:235 channel_on_init()
Ring-Ready portaudio/1003!
*pa answer*
2008-12-11 14:17:24 [NOTICE] mod_portaudio.c:1404 answer_call() Channel
[portaudio/1003] has been answered
API CALL [pa(answer)] output:
Answered 1 channels.

freeswitch at localhost> 2008-12-11 14:17:24 [NOTICE]
switch_ivr_originate.c:1509 switch_ivr_originate() Channel [sofia/internal/
1003 at 172.20.177.117] has been answered
pa hangup
2008-12-11 14:17:32 [NOTICE] mod_portaudio.c:1365 hangup_call() Hangup
portaudio/1003 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
API CALL [pa(hangup)] output:
OK

freeswitch at localhost> 2008-12-11 14:17:32 [NOTICE]
switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup
sofia/internal/1003 at 172.20.177.117 [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-11 14:17:32 [INFO] mod_cdr_csv.c:207 my_on_hangup() CHANNEL_DATA:
Channel-State: [CS_HANGUP]
Channel-State-Number: [10]
Channel-Name: [sofia/internal/1003 at 172.20.177.117]
Unique-ID: [d49468e5-be90-40f6-8ffe-58c56651d87a]
Call-Direction: [inbound]
Answer-State: [answered]
Caller-Username: [1003]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [FreeSwitch]
Caller-Caller-ID-Number: [1003]
Caller-Network-Addr: [172.20.177.201]
Caller-Destination-Number: [3]
Caller-Unique-ID: [d49468e5-be90-40f6-8ffe-58c56651d87a]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/1003 at 172.20.177.117]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1228985236919311]
Caller-Channel-Created-Time: [1228985236919311]
Caller-Channel-Answered-Time: [1228985244720686]
Caller-Channel-Progress-Time: [1228985237539254]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [1228985252428470]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
Other-Leg-Username: [1003]
Other-Leg-Dialplan: [XML]
Other-Leg-Caller-ID-Name: [Extension 1003]
Other-Leg-Caller-ID-Number: [1003]
Other-Leg-Network-Addr: [172.20.177.201]
Other-Leg-Unique-ID: [bfc0023c-0725-40a8-a187-574bdab40c40]
Other-Leg-Source: [mod_sofia]
Other-Leg-Context: [default]
Other-Leg-Channel-Name: [portaudio/1003]
Other-Leg-Screen-Bit: [true]
Other-Leg-Privacy-Hide-Name: [false]
Other-Leg-Privacy-Hide-Number: [false]


2008-12-11 14:17:32 [NOTICE] switch_core_session.c:956
switch_core_session_thread() Session 1 (sofia/internal/1003 at 172.20.177.117)
Ended
2008-12-11 14:17:32 [NOTICE] switch_core_session.c:958
switch_core_session_thread() Close Channel sofia/internal/
1003 at 172.20.177.117 [CS_HANGUP]
2008-12-11 14:17:32 [NOTICE] switch_core_session.c:956
switch_core_session_thread() Session 2 (portaudio/1003) Ended
2008-12-11 14:17:32 [NOTICE] switch_core_session.c:958
switch_core_session_thread() Close Channel portaudio/1003 [CS_HANGUP]
*
Then i tried it for outbound
Output:*

*pa call 1003*
2008-12-11 14:17:39 [NOTICE] switch_channel.c:564 switch_channel_set_name()
New Channel portaudio/1003 [14de48f4-40cb-42bb-8f43-084d4df7ec89]
2008-12-11 14:17:39 [NOTICE] mod_portaudio.c:1586 place_call() Channel
[portaudio/1003] has been answered
API CALL [pa(call 1003)] output:
SUCCESS:2:14de48f4-40cb-42bb-8f43-084d4df7ec89

freeswitch at localhost> 2008-12-11 14:17:39 [INFO] mod_dialplan_xml.c:233
dialplan_hunt() Processing FreeSWITCH->1003 in context default
2008-12-11 14:17:39 [ERR] mod_sofia.c:2102 sofia_outgoing_channel() Invalid
Gateway
2008-12-11 14:17:39 [NOTICE] mod_sofia.c:2301 sofia_outgoing_channel() Close
Channel N/A [CS_NEW]
2008-12-11 14:17:39 [ERR] switch_ivr_originate.c:1063 switch_ivr_originate()
Cannot create outgoing channel of type [sofia] cause:
[INVALID_NUMBER_FORMAT]
2008-12-11 14:17:39 [INFO] mod_dptools.c:1868 audio_bridge_function()
Originate Failed.  Cause: INVALID_NUMBER_FORMAT
2008-12-11 14:17:39 [NOTICE] mod_dptools.c:1895 audio_bridge_function()
Hangup portaudio/1003 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2008-12-11 14:17:39 [INFO] mod_cdr_csv.c:207 my_on_hangup() CHANNEL_DATA:
Channel-State: [CS_HANGUP]
Channel-State-Number: [10]
Channel-Name: [portaudio/1003]
Unique-ID: [14de48f4-40cb-42bb-8f43-084d4df7ec89]
Call-Direction: [inbound]
Answer-State: [answered]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [FreeSWITCH]
Caller-Caller-ID-Number: [0000000000]
Caller-Network-Addr: [172.20.177.117]
Caller-Destination-Number: [1003]
Caller-Unique-ID: [14de48f4-40cb-42bb-8f43-084d4df7ec89]
Caller-Source: [mod_portaudio]
Caller-Context: [default]
Caller-Channel-Name: [portaudio/1003]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1228985259280611]
Caller-Channel-Created-Time: [1228985259280611]
Caller-Channel-Answered-Time: [1228985259508649]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [1228985259512660]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]


2008-12-11 14:17:39 [NOTICE] switch_core_session.c:956
switch_core_session_thread() Session 3 (portaudio/1003) Ended
2008-12-11 14:17:39 [NOTICE] switch_core_session.c:958
switch_core_session_thread() Close Channel portaudio/1003 [CS_HANGUP]

*Correct me were in am wrong . i have done all the updates and i install
freeswitch newly .

I am using Centos 5.2

I also attached the default.xml in this mail.

Correct me were in am wrong.

-- 
Thanks,
N.Baskar*
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From carole.olivier at enst.fr  Thu Dec 11 03:14:12 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 11 Dec 2008 03:14:12 -0800 (PST)
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <7DF34003-ACE0-496A-B986-E91E83C75C4B@freeswitch.org>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
	<87f2f3b90812102305o44a5d2f3x11b900f86c0a4242@mail.gmail.com>
	<20950711.post@talk.nabble.com>
	<7DF34003-ACE0-496A-B986-E91E83C75C4B@freeswitch.org>
Message-ID: <20953570.post@talk.nabble.com>


ok !

Well, I installed it from opensuse.org, I thought it would be easier for me
since I am completely new here.
Is there a simple way to update this package or would you recommend me to
uninstall the rpm and install freeswitch completely new from the source code
you provide?

Thanks,
Carole


Michael Collins-11 wrote:
> 
> Wow! That is really old. I strongly recommend that you update to the  
> latest trunk using svn.
> 
> -MC
> 
> 
> On Dec 10, 2008, at 11:33 PM, "Carole O." <carole.olivier at enst.fr>  
> wrote:
> 
>>
>> I have got the following:
>> Freeswitch Version 1.0.1 (9171)
>>
>> Carole
>>
>>
>>
>> Michael Collins-11 wrote:
>>>
>>> Carole,
>>>
>>> There have been many updates since 1.0.1 was officially released. If
>>> you could start FreeSWITCH and then press F12 it will reveal which  
>>> SVN
>>> revisionnumber you are running. Please supply that number and it will
>>> help us to know if you are on a recent revision.
>>>
>>> Thanks,
>>> MC
>>>
>>> On Wed, Dec 10, 2008 at 10:51 PM, Carole O. <carole.olivier at enst.fr>
>>> wrote:
>>>>
>>>> Hello,
>>>>
>>>> I am running the version 1.0.1.
>>>> Do you still need me to run the debugging?
>>>>
>>>> Carole
>>>>
>>>>
>>>> Brian West-3 wrote:
>>>>>
>>>>> I have already labbed this up on SVN trunk and I don't get a  
>>>>> segfault
>>>>> but I get something else that prevents it from working  
>>>>> properly.    We
>>>>> are working on it today.  Also what version are you running?
>>>>>
>>>>> /b
>>>>>
>>>>> On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
>>>>>
>>>>>> Thanks for reporting this. It would be helpful to know a bit  
>>>>>> more. Can
>>>>>> you start freeswitch and press F12 (or type "version" at the  
>>>>>> CLI) and
>>>>>> report back what it says?
>>>>>> Also, a backtrace (bt) is generally useful. If you could produce a
>>>>>> "bt" and a "bt full" from you core file that would be extremely
>>>>>> helpful.
>>>>>>
>>>>>> see this link for more information:
>>>>>> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>>>>
>>>>>> you should have a "core" file for each segfault that occurred.  
>>>>>> Use the
>>>>>> gdb program to get the back trace:
>>>>>>
>>>>>> gdb /path/to/fs/binary core.xxx
>>>>>>
>>>>>> then capture the output from these two commands:
>>>>>>
>>>>>> bt <enter>
>>>>>> bt full <enter>
>>>>>>
>>>>>> When you type those commands you'll see tons of debugging info;
>>>>>> capture that and put it in a pastebin (pastebin.freeswitch.org)  
>>>>>> then
>>>>>> report back here.
>>>>>>
>>>>>> You can exit the gdb debugger by typing q <enter>
>>>>>>
>>>>>> Thanks for helping us collect information!
>>>>>>
>>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> --
>>>> View this message in context:
>>>> http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950299.html
>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> -- 
>> View this message in context:
>> http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20950711.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20953570.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From william at channelxstream.com  Wed Dec 10 23:01:02 2008
From: william at channelxstream.com (William King)
Date: Wed, 10 Dec 2008 23:01:02 -0800
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <20950299.post@talk.nabble.com>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
Message-ID: <1228978862.31447.5.camel@quentusrex-desktop>

Carole,

1.0.1 is known to be broken now. Can you go into the directory where you
have the freeswitch source from the svn repo and type 'make current'.
This will update freeswitch to trunk.

-William 

On Wed, 2008-12-10 at 22:51 -0800, Carole O. wrote:
> Hello,
> 
> I am running the version 1.0.1.
> Do you still need me to run the debugging?
> 
> Carole
> 
> 
> Brian West-3 wrote:
> > 
> > I have already labbed this up on SVN trunk and I don't get a segfault  
> > but I get something else that prevents it from working properly.    We  
> > are working on it today.  Also what version are you running?
> > 
> > /b
> > 
> > On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
> > 
> >> Thanks for reporting this. It would be helpful to know a bit more. Can
> >> you start freeswitch and press F12 (or type "version" at the CLI) and
> >> report back what it says?
> >> Also, a backtrace (bt) is generally useful. If you could produce a
> >> "bt" and a "bt full" from you core file that would be extremely
> >> helpful.
> >>
> >> see this link for more information:
> >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
> >>
> >> you should have a "core" file for each segfault that occurred. Use the
> >> gdb program to get the back trace:
> >>
> >> gdb /path/to/fs/binary core.xxx
> >>
> >> then capture the output from these two commands:
> >>
> >> bt <enter>
> >> bt full <enter>
> >>
> >> When you type those commands you'll see tons of debugging info;
> >> capture that and put it in a pastebin (pastebin.freeswitch.org) then
> >> report back here.
> >>
> >> You can exit the gdb debugger by typing q <enter>
> >>
> >> Thanks for helping us collect information!
> >>
> >> -MC
> > 
> > 
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> 
-- 
William King
Cell: 253-686-5518
E-mail: william at channelxstream.com
Channel XStream
www.channelxstream.com
1-877-600-6786

If there is a possibility that any information in our conversation might
be considered 'private' or 'sensitive' such as passwords, account 
information, legal or financial information, or anything else that you
would consider 'private' or 'sensitive' communications. It is better to 
always err on the side of security. Please encrypt the e-mail using 
my gpg key: 95C9D5B3.

If you are unfamiliar with e-mail encryption feel free to let me know 
and I can help you establish the proper protocols and procedures.
https://help.ubuntu.com/community/GnuPrivacyGuardHowto

Get my gpg key: gpg --recv-key --keyserver keyserver.ubuntu.com 95C9D5B3
Key Fingerprint: EA6F B2EE 1846 55D4 FFD9  80BA 6489 B48C 95C9 D5B3
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From alex at sinapticode.ro  Thu Dec 11 01:21:07 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 11:21:07 +0200
Subject: [Freeswitch-users] Configuring FreeSwitch
Message-ID: <1228987267.4843.6.camel@gathern.lan>

Hi,

I'm a newbie trying to configure freeswitch.

Our needs are simple: for starters we need to call external phones from
freeswitch and play a wav file. 

Now, we have the voice provider's IP (no user/password authentication),
and we have the codec used (called "ulaw" in Asterisk, which I think is
called PCMU?). And this is basically all that we specify in Asterisk.

I compiled Freeswitch with the default settings, and I was wondering if
you guys can give me some hits, point me in the right direction.

Thanks,

-- 
Alexandru Nedelcu
http://alexn.org




From mike at jerris.com  Thu Dec 11 03:32:37 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Dec 2008 06:32:37 -0500
Subject: [Freeswitch-users] Console Dialing in Freeswitch
In-Reply-To: <d0640d380812110051w41bc5481r60a3d3242044d3f7@mail.gmail.com>
References: <d0640d380811272338o35e36cd4ta1f2296449d300ed@mail.gmail.com>
	<7b197bef0812012250n7173710fic38fa04154b1a40b@mail.gmail.com>
	<d0640d380812012327y2ca9bfbbjf75b9fe95b622e1@mail.gmail.com>
	<7b197bef0812012339x376145d8i7e2a27546ec937c8@mail.gmail.com>
	<d0640d380812020257m2a135790u3ea4cc2ecc8d85cc@mail.gmail.com>
	<59157E68-806F-4038-8CFD-D35C54D711F9@jerris.com>
	<d0640d380812020307h5a6e00dcladf141c45b716176@mail.gmail.com>
	<191c3a030812020809m428b656av261ec5bf1ad102@mail.gmail.com>
	<d0640d380812030732l2d98dd7co36ac69d96e2c643c@mail.gmail.com>
	<191c3a030812030924u457f934ep77bd70680f583fcd@mail.gmail.com>
	<d0640d380812110051w41bc5481r60a3d3242044d3f7@mail.gmail.com>
Message-ID: <4A5A4648-37F4-4887-AD71-005998A7E8CD@jerris.com>

Check your dialplan, you have a gateway name that is not configured  
properly.


On Dec 11, 2008, at 3:51 AM, Baskar wrote:

> 2008-12-11 14:17:39 [ERR] mod_sofia.c:2102 sofia_outgoing_channel()  
> Invalid Gateway




From alex at sinapticode.ro  Thu Dec 11 03:55:39 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 13:55:39 +0200
Subject: [Freeswitch-users] Freeswitch Dialer configuration
Message-ID: <1228996539.4843.13.camel@gathern.lan>

Hi,

I'm trying to setup a simple dialer with Freeswitch.

What I have right now is the following dialplan:

  <extension name="wavs">
    <condition field="destination_number" expression="^2009$">
      <action application="playback"
data="/var/lib/axonite/factura.wav"/>
    </condition>
  </extension>

Right now I'm testing the setup with the following command:
  originate sofia/external/123123123123 at provider.com 2009

How can I configure it (or where to find examples) for the following:
1) the message should start when the phone is answered (right now it
starts when the phone starts ringing I think)
2) I need keys interaction ... like when the user presses 1, the message
should repeat itself, and when the user presses 2 another message should
play

Thank you,

-- 
Alexandru Nedelcu
Software Developer, Sinapticode




From carole.olivier at enst.fr  Thu Dec 11 05:13:47 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 11 Dec 2008 05:13:47 -0800 (PST)
Subject: [Freeswitch-users]  conference_auto_outcall_announce
Message-ID: <20955216.post@talk.nabble.com>


Hello,

I would like to call automatically a group of user and make them listening
an announcement before anybody talks. I have already added an extension in
the dialplan that worked. Actually, I use the extension 0911 in the default
dialplan of freeswitch which I have changed a little. This works fine. The
problem is to make the announcement.

1-
I have created a profile for this conference and add:
 < param name="enter-sound" value="conference/my_file.wav"/> 

It did not work. I have tested this, it seems to work only if someone enters
a conference which has been previously created. For instance after the 1st
participant has entered the conference room.


2- 
Then, I have seen there is the function "conference_auto_outcall_announce"
but I did not manage to make it work.
I have added in my extension:
<action application="set"
data="conference_auto_outcall_announce=play:path_file_to_play/file.wav"/>

I have got the following error:
[ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File
[/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
[System error : no such file or directory]

I have tried to change the path but something else is wrong.


3-
I have tried with the API command:
conference <confname> play <file_path>
It works from the console but not from the dialpan.


If someone could tell me where I am wrong or has another idea it would be
very helpfull.

Thanks,
Carole



-- 
View this message in context: http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20955216.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Thu Dec 11 05:57:52 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Dec 2008 07:57:52 -0600
Subject: [Freeswitch-users] conference_auto_outcall_announce
In-Reply-To: <20955216.post@talk.nabble.com>
References: <20955216.post@talk.nabble.com>
Message-ID: <0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>

Don't have play: in there and it should be fine.  Also if you want the  
absolute path you start it with /path/to/file.wav


/b

On Dec 11, 2008, at 7:13 AM, Carole O. wrote:

> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
> [System error : no such file or directory]




From john at loopfx.com  Thu Dec 11 06:29:58 2008
From: john at loopfx.com (John Rutherford)
Date: Thu, 11 Dec 2008 09:29:58 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local><87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local><49402977.4090808@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>
	<87f2f3b90812101452t7076bb6fo93f7a78bbfb0404f@mail.gmail.com>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360DB5@anniesue.loop.local>

No.  I wish it were that simple.  

I'm doing all of my testing on an internal network.  

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Wednesday, December 10, 2008 5:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

I smell a NAT... is there any NAT involved?

On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford <john at loopfx.com>
wrote:
> Okay. I just tried this.
>
>
>
> Now we're getting the audio going one way, but not the other.  So, I
can
> hear the person that I just transferred to, but they can't hear me.
>
>
>
> Anyone have any other ideas?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Raymond
> Chandler
> Sent: Wednesday, December 10, 2008 3:41 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> try blocking ICMP packets TO the MSS.... i had this exact same problem
a few
> months ago.... MSS starts sending RTP to FS before FS is ready to
accept....
> so the OS catches the port not open and returns an ICMP 3:3 back to
the
> MSS.... which in turn chokes on the queued up RTP and refuses to send
> anymore...
>
> -Ray
>
> John Rutherford wrote:
>
> I just emailed it to him.
>
>
>
> Thanks!
>
>
>
> -----Original Message-----
>
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>
> Michael Collins
>
> Sent: Wednesday, December 10, 2008 1:09 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
> On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
>
> wrote:
>
>
>
> I have a pcap, but I'm not able to see anything obviously wrong with
>
>
>
> it.
>
>
>
> We find that some equipment (in fact a lot of equipment) have features
>
> that cause issues to be quite non-obvious, so perhaps you could give
>
> the pcap to Brian for him to review. He's a total ace when it comes to
>
> bug hunting.
>
>
>
> -MC
>
>
>
>
>
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>
>
>
> Brian
>
>
>
> West
>
> Sent: Wednesday, December 10, 2008 12:59 PM
>
> To: freeswitch-users at lists.freeswitch.org
>
> Subject: Re: [Freeswitch-users] No audio after transfer
>
>
>
>
>
>
>
> would be most helpful to capture a pcap of the entire thing by itself
>
>
>
> start
>
>
>
> to finish.
>
>
>
>
>
>
>
> /b
>
>
>
>
>
>
>
> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>
>
>
> No. I realize that's it's a B2BUA and that's exactly what we want.
>
>
>
>
>
>
>
> Everything with the transfer seems to work fine, except that there is
>
>
>
> no
>
>
>
> audio.
>
>
>
>
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From carole.olivier at enst.fr  Thu Dec 11 06:36:48 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 11 Dec 2008 06:36:48 -0800 (PST)
Subject: [Freeswitch-users] conference_auto_outcall_announce
In-Reply-To: <0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>
References: <20955216.post@talk.nabble.com>
	<0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>
Message-ID: <20956587.post@talk.nabble.com>


Hello,

Actually, I have already tried it but nothing happens: the file is not
played and there is no error. 
There is still a difference: if I configure it as you said, I can not be
listening anymore, there is simply nothing.

Would you have an idea? I have checked the path and the syntax 1 million
times so I do not think I make mistake there.

Thanks,
Carole



Brian West-3 wrote:
> 
> Don't have play: in there and it should be fine.  Also if you want the  
> absolute path you start it with /path/to/file.wav
> 
> 
> /b
> 
> On Dec 11, 2008, at 7:13 AM, Carole O. wrote:
> 
>> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
>> [System error : no such file or directory]
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20956587.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From intralanman at freeswitch.org  Thu Dec 11 06:39:11 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 11 Dec 2008 09:39:11 -0500
Subject: [Freeswitch-users] No audio after transfer
In-Reply-To: <81469655CA61444CBB034826ABC6F6E3360DB5@anniesue.loop.local>
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local><87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local><49402977.4090808@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>	<87f2f3b90812101452t7076bb6fo93f7a78bbfb0404f@mail.gmail.com>
	<81469655CA61444CBB034826ABC6F6E3360DB5@anniesue.loop.local>
Message-ID: <4941260F.90704@freeswitch.org>

can you send a pcap of sip and rtp with the new problem?

-Ray


John Rutherford wrote:
> No.  I wish it were that simple.  
>
> I'm doing all of my testing on an internal network.  
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
> Sent: Wednesday, December 10, 2008 5:53 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] No audio after transfer
>
> I smell a NAT... is there any NAT involved?
>
> On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford <john at loopfx.com>
> wrote:
>   
>> Okay. I just tried this.
>>
>>
>>
>> Now we're getting the audio going one way, but not the other.  So, I
>>     
> can
>   
>> hear the person that I just transferred to, but they can't hear me.
>>
>>
>>
>> Anyone have any other ideas?
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>     
> Raymond
>   
>> Chandler
>> Sent: Wednesday, December 10, 2008 3:41 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] No audio after transfer
>>
>>
>>
>> try blocking ICMP packets TO the MSS.... i had this exact same problem
>>     
> a few
>   
>> months ago.... MSS starts sending RTP to FS before FS is ready to
>>     
> accept....
>   
>> so the OS catches the port not open and returns an ICMP 3:3 back to
>>     
> the
>   
>> MSS.... which in turn chokes on the queued up RTP and refuses to send
>> anymore...
>>
>> -Ray
>>
>> John Rutherford wrote:
>>
>> I just emailed it to him.
>>
>>
>>
>> Thanks!
>>
>>
>>
>> -----Original Message-----
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>>
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>
>> Michael Collins
>>
>> Sent: Wednesday, December 10, 2008 1:09 PM
>>
>> To: freeswitch-users at lists.freeswitch.org
>>
>> Subject: Re: [Freeswitch-users] No audio after transfer
>>
>>
>>
>> On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford <john at loopfx.com>
>>
>> wrote:
>>
>>
>>
>> I have a pcap, but I'm not able to see anything obviously wrong with
>>
>>
>>
>> it.
>>
>>
>>
>> We find that some equipment (in fact a lot of equipment) have features
>>
>> that cause issues to be quite non-obvious, so perhaps you could give
>>
>> the pcap to Brian for him to review. He's a total ace when it comes to
>>
>> bug hunting.
>>
>>
>>
>> -MC
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>>
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>
>>
>>
>> Brian
>>
>>
>>
>> West
>>
>> Sent: Wednesday, December 10, 2008 12:59 PM
>>
>> To: freeswitch-users at lists.freeswitch.org
>>
>> Subject: Re: [Freeswitch-users] No audio after transfer
>>
>>
>>
>>
>>
>>
>>
>> would be most helpful to capture a pcap of the entire thing by itself
>>
>>
>>
>> start
>>
>>
>>
>> to finish.
>>
>>
>>
>>
>>
>>
>>
>> /b
>>
>>
>>
>>
>>
>>
>>
>> On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
>>
>>
>>
>> No. I realize that's it's a B2BUA and that's exactly what we want.
>>
>>
>>
>>
>>
>>
>>
>> Everything with the transfer seems to work fine, except that there is
>>
>>
>>
>> no
>>
>>
>>
>> audio.
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>> Freeswitch-users mailing list
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>>
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>> Freeswitch-users mailing list
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>>
>> Freeswitch-users mailing list
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>     
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From intralanman at freeswitch.org  Thu Dec 11 06:55:30 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 11 Dec 2008 09:55:30 -0500
Subject: [Freeswitch-users] Configuring FreeSwitch
In-Reply-To: <1228987267.4843.6.camel@gathern.lan>
References: <1228987267.4843.6.camel@gathern.lan>
Message-ID: <494129E2.5010602@freeswitch.org>

Alexandru Nedelcu wrote:
> Hi,
>
> I'm a newbie trying to configure freeswitch.
>
> Our needs are simple: for starters we need to call external phones from
> freeswitch and play a wav file. 
>
> Now, we have the voice provider's IP (no user/password authentication),
> and we have the codec used (called "ulaw" in Asterisk, which I think is
> called PCMU?). And this is basically all that we specify in Asterisk.
>
>   
i think i answered all of this for you on irc yesterday....

use the bridge dialplan app to dial by ip similar to the following:
    <action application="bridge" 
data="sofia/${use_profile}/number at ip.address"/>

http://wiki.freeswitch.org/wiki/Sofia#Syntax might also help you out a 
little




From alex at sinapticode.ro  Thu Dec 11 07:25:01 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 17:25:01 +0200
Subject: [Freeswitch-users] Having trouble with capturing key presses from
	Javascript
Message-ID: <1229009101.4843.18.camel@gathern.lan>

Hi,

I've tried a simple example from here:
http://wiki.freeswitch.org/wiki/Session_streamFile
You can view my code here:
http://paste.scsys.co.uk/paste

The problem is, streamFile doesn't fire the event on keypress.
Is there anything wrong with my code?

Thank you,

-- 
Alexandru Nedelcu
Software Developer, Sinapticode




From alex at sinapticode.ro  Thu Dec 11 07:59:14 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 17:59:14 +0200
Subject: [Freeswitch-users] Having trouble with capturing key presses
	from Javascript
In-Reply-To: <1229009101.4843.18.camel@gathern.lan>
References: <1229009101.4843.18.camel@gathern.lan>
Message-ID: <1229011154.4843.21.camel@gathern.lan>

On Thu, 2008-12-11 at 17:25 +0200, Alexandru Nedelcu wrote:
> Hi,
> 
> I've tried a simple example from here:
> http://wiki.freeswitch.org/wiki/Session_streamFile
> You can view my code here:
> http://paste.scsys.co.uk/paste

Sorry, the correct link was:
http://scsys.co.uk:8001/21364

> 
> The problem is, streamFile doesn't fire the event on keypress.
> Is there anything wrong with my code?

Another thing: in Asterisk I had the following setting in sip.conf ...
dtmfmode=inband
Without it key presses wasn't working in Asterisk.

Is there something similar for Freeswitch?




From intralanman at freeswitch.org  Thu Dec 11 08:22:11 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 11 Dec 2008 11:22:11 -0500
Subject: [Freeswitch-users] Having trouble with capturing key presses
 from Javascript
In-Reply-To: <1229011154.4843.21.camel@gathern.lan>
References: <1229009101.4843.18.camel@gathern.lan>
	<1229011154.4843.21.camel@gathern.lan>
Message-ID: <49413E33.2030604@freeswitch.org>

use the start_dtmf app to get inband dtmf

-Ray

Alexandru Nedelcu wrote:
> On Thu, 2008-12-11 at 17:25 +0200, Alexandru Nedelcu wrote:
>   
>> Hi,
>>
>> I've tried a simple example from here:
>> http://wiki.freeswitch.org/wiki/Session_streamFile
>> You can view my code here:
>> http://paste.scsys.co.uk/paste
>>     
>
> Sorry, the correct link was:
> http://scsys.co.uk:8001/21364
>
>   
>> The problem is, streamFile doesn't fire the event on keypress.
>> Is there anything wrong with my code?
>>     
>
> Another thing: in Asterisk I had the following setting in sip.conf ...
> dtmfmode=inband
> Without it key presses wasn't working in Asterisk.
>
> Is there something similar for Freeswitch?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From alex at sinapticode.ro  Thu Dec 11 08:29:35 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Thu, 11 Dec 2008 18:29:35 +0200
Subject: [Freeswitch-users] Having trouble with capturing key
	presses	from Javascript
In-Reply-To: <49413E33.2030604@freeswitch.org>
References: <1229009101.4843.18.camel@gathern.lan>
	<1229011154.4843.21.camel@gathern.lan>
	<49413E33.2030604@freeswitch.org>
Message-ID: <1229012975.4843.23.camel@gathern.lan>

Can the app be initiated from JS?


On Thu, 2008-12-11 at 11:22 -0500, Raymond Chandler wrote:
> use the start_dtmf app to get inband dtmf
> 
> -Ray
> 
> Alexandru Nedelcu wrote: 
> > On Thu, 2008-12-11 at 17:25 +0200, Alexandru Nedelcu wrote:
> >   
> > > Hi,
> > > 
> > > I've tried a simple example from here:
> > > http://wiki.freeswitch.org/wiki/Session_streamFile
> > > You can view my code here:
> > > http://paste.scsys.co.uk/paste
> > >     
> > 
> > Sorry, the correct link was:
> > http://scsys.co.uk:8001/21364
> > 
> >   
> > > The problem is, streamFile doesn't fire the event on keypress.
> > > Is there anything wrong with my code?
> > >     
> > 
> > Another thing: in Asterisk I had the following setting in sip.conf ...
> > dtmfmode=inband
> > Without it key presses wasn't working in Asterisk.
> > 
> > Is there something similar for Freeswitch?
> > 
> > 
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >   
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Thu Dec 11 08:39:48 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Dec 2008 10:39:48 -0600
Subject: [Freeswitch-users] Having trouble with capturing key
	presses	from Javascript
In-Reply-To: <1229012975.4843.23.camel@gathern.lan>
References: <1229009101.4843.18.camel@gathern.lan>
	<1229011154.4843.21.camel@gathern.lan>
	<49413E33.2030604@freeswitch.org>
	<1229012975.4843.23.camel@gathern.lan>
Message-ID: <039F34E7-5CFF-4185-A29A-FB62BC6A28AE@freeswitch.org>

session.execute("start_dtmf")

/b

On Dec 11, 2008, at 10:29 AM, Alexandru Nedelcu wrote:

> Can the app be initiated from JS?




From john at loopfx.com  Thu Dec 11 10:42:54 2008
From: john at loopfx.com (John Rutherford)
Date: Thu, 11 Dec 2008 13:42:54 -0500
Subject: [Freeswitch-users] No audio after transfer
References: <81469655CA61444CBB034826ABC6F6E331D817@anniesue.loop.local><191c3a030812100734y2f1bf9ds492ec06b49dfe3b@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C32@anniesue.loop.local><BA9AB06E-F63F-4262-B1ED-556B4AC185F3@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360C37@anniesue.loop.local><87f2f3b90812101008t3fd3f4day821330cc60662a72@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360C72@anniesue.loop.local><49402977.4090808@freeswitch.org><81469655CA61444CBB034826ABC6F6E3360D31@anniesue.loop.local>	<87f2f3b90812101452t7076bb6fo93f7a78bbfb0404f@mail.gmail.com><81469655CA61444CBB034826ABC6F6E3360DB5@anniesue.loop.local>
	<4941260F.90704@freeswitch.org>
Message-ID: <81469655CA61444CBB034826ABC6F6E3360E73@anniesue.loop.local>

Sent.  

 

Let me know if you see anything. I'm not able to see anything wrong.

 

Thanks,

John

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Raymond Chandler
Sent: Thursday, December 11, 2008 9:39 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

can you send a pcap of sip and rtp with the new problem?

-Ray 


John Rutherford wrote: 

No.  I wish it were that simple.  
 
I'm doing all of my testing on an internal network.  
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Wednesday, December 10, 2008 5:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer
 
I smell a NAT... is there any NAT involved?
 
On Wed, Dec 10, 2008 at 2:18 PM, John Rutherford <john at loopfx.com>
<mailto:john at loopfx.com> 
wrote:
  

	Okay. I just tried this.
	 
	 
	 
	Now we're getting the audio going one way, but not the other.
So, I
	    

can
  

	hear the person that I just transferred to, but they can't hear
me.
	 
	 
	 
	Anyone have any other ideas?
	 
	 
	 
	From: freeswitch-users-bounces at lists.freeswitch.org
	[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
	    

Raymond
  

	Chandler
	Sent: Wednesday, December 10, 2008 3:41 PM
	To: freeswitch-users at lists.freeswitch.org
	Subject: Re: [Freeswitch-users] No audio after transfer
	 
	 
	 
	try blocking ICMP packets TO the MSS.... i had this exact same
problem
	    

a few
  

	months ago.... MSS starts sending RTP to FS before FS is ready
to
	    

accept....
  

	so the OS catches the port not open and returns an ICMP 3:3 back
to
	    

the
  

	MSS.... which in turn chokes on the queued up RTP and refuses to
send
	anymore...
	 
	-Ray
	 
	John Rutherford wrote:
	 
	I just emailed it to him.
	 
	 
	 
	Thanks!
	 
	 
	 
	-----Original Message-----
	 
	From: freeswitch-users-bounces at lists.freeswitch.org
	 
	[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
	 
	Michael Collins
	 
	Sent: Wednesday, December 10, 2008 1:09 PM
	 
	To: freeswitch-users at lists.freeswitch.org
	 
	Subject: Re: [Freeswitch-users] No audio after transfer
	 
	 
	 
	On Wed, Dec 10, 2008 at 10:01 AM, John Rutherford
<john at loopfx.com> <mailto:john at loopfx.com> 
	 
	wrote:
	 
	 
	 
	I have a pcap, but I'm not able to see anything obviously wrong
with
	 
	 
	 
	it.
	 
	 
	 
	We find that some equipment (in fact a lot of equipment) have
features
	 
	that cause issues to be quite non-obvious, so perhaps you could
give
	 
	the pcap to Brian for him to review. He's a total ace when it
comes to
	 
	bug hunting.
	 
	 
	 
	-MC
	 
	 
	 
	 
	 
	 
	 
	 
	 
	From: freeswitch-users-bounces at lists.freeswitch.org
	 
	[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
	 
	 
	 
	Brian
	 
	 
	 
	West
	 
	Sent: Wednesday, December 10, 2008 12:59 PM
	 
	To: freeswitch-users at lists.freeswitch.org
	 
	Subject: Re: [Freeswitch-users] No audio after transfer
	 
	 
	 
	 
	 
	 
	 
	would be most helpful to capture a pcap of the entire thing by
itself
	 
	 
	 
	start
	 
	 
	 
	to finish.
	 
	 
	 
	 
	 
	 
	 
	/b
	 
	 
	 
	 
	 
	 
	 
	On Dec 10, 2008, at 11:51 AM, John Rutherford wrote:
	 
	 
	 
	No. I realize that's it's a B2BUA and that's exactly what we
want.
	 
	 
	 
	 
	 
	 
	 
	Everything with the transfer seems to work fine, except that
there is
	 
	 
	 
	no
	 
	 
	 
	audio.
	 
	 
	 
	 
	 
	 
	 
	_______________________________________________
	 
	Freeswitch-users mailing list
	 
	Freeswitch-users at lists.freeswitch.org
	 
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	 
	 
	 
	 
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	 
	 
	http://www.freeswitch.org
	 
	 
	 
	 
	 
	 
	 
	 
	 
	_______________________________________________
	 
	Freeswitch-users mailing list
	 
	Freeswitch-users at lists.freeswitch.org
	 
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	http://www.freeswitch.org
	 
	 
	 
	_______________________________________________
	 
	Freeswitch-users mailing list
	 
	Freeswitch-users at lists.freeswitch.org
	 
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	http://www.freeswitch.org
	 
	 
	 
	 
	 
	_______________________________________________
	Freeswitch-users mailing list
	Freeswitch-users at lists.freeswitch.org
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
	 
	    

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  

	http://www.freeswitch.org
	 
	 
	    

 
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  

 

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From vkobashi at ydeasolutions.com.br  Thu Dec 11 11:59:31 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Thu, 11 Dec 2008 16:59:31 -0300
Subject: [Freeswitch-users]  LDAP Integration
Message-ID: <49417123.10709@ydeasolutions.com.br>

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From vkobashi at ydeasolutions.com.br  Thu Dec 11 12:16:56 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Thu, 11 Dec 2008 17:16:56 -0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49417123.10709@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>
Message-ID: <49417538.9040203@ydeasolutions.com.br>

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From hads at nice.net.nz  Thu Dec 11 11:42:00 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Fri, 12 Dec 2008 08:42:00 +1300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49417538.9040203@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>
	<49417538.9040203@ydeasolutions.com.br>
Message-ID: <200812120842.00808.hads@nice.net.nz>

On Friday 12 December 2008 09:16:56 Vinicius Kobashi wrote:
> i found another module called mod_xml_curl and loaded it to freeswitch
> too... but still it shows me the following error:
>
>  2008-12-11 17:04:04 [WARNING] sofia_reg.c:1501 sofia_reg_parse_auth()
> Can't find user [username at freeswitchserver.com] You must define a domain
> called 'freeswitchserver.com' in your directory and add a user with the
> id="username" attribute and you must configure your device to use the
> proper domain in it's authentication credentials.
>
>  does anyone got an idea?

Yes, you need to define a domain called 'freeswitchserver.com' in your 
directory and add a user with the id="username" just like the error message 
says.

The directory files are in conf/directory/

If you would like to read up on mod_xml_curl there is a detailed page on the 
wiki;

http://wiki.freeswitch.org/wiki/Mod_xml_curl

hads
-- 
http://nicegear.co.nz
VoIP, DVB and other Linux compatible hardware.



From vkobashi at ydeasolutions.com.br  Thu Dec 11 13:35:12 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Thu, 11 Dec 2008 18:35:12 -0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <200812120842.00808.hads@nice.net.nz>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>
	<200812120842.00808.hads@nice.net.nz>
Message-ID: <49418790.60001@ydeasolutions.com.br>

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From brian at freeswitch.org  Thu Dec 11 12:40:18 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Dec 2008 14:40:18 -0600
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49418790.60001@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>
	<200812120842.00808.hads@nice.net.nz>
	<49418790.60001@ydeasolutions.com.br>
Message-ID: <2F2BD503-9199-4131-998A-3777809624CD@freeswitch.org>

Thats already fixed too.. update.

/b

On Dec 11, 2008, at 3:35 PM, Vinicius Kobashi wrote:

> ok ill try that
>
> i found another module thats mod_xml_ldap
>
> but when i try to load it, during compiling i get the 404 error http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz 
>  file not found....
> ill try to download it myself and then try to compile freeswitch  
> again and test
>
> =D thankz for the fast answer

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From msc at freeswitch.org  Thu Dec 11 12:41:56 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Dec 2008 12:41:56 -0800
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49418790.60001@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>
	<49417538.9040203@ydeasolutions.com.br>
	<200812120842.00808.hads@nice.net.nz>
	<49418790.60001@ydeasolutions.com.br>
Message-ID: <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>

Please confirm your svn rev - I believe this was fixed recently. Do
"make current" in your source directory.
-MC

On Thu, Dec 11, 2008 at 1:35 PM, Vinicius Kobashi
<vkobashi at ydeasolutions.com.br> wrote:
> ok ill try that
>
> i found another module thats mod_xml_ldap
>
> but when i try to load it, during compiling i get the 404 error
> http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz file not
> found....
> ill try to download it myself and then try to compile freeswitch again and
> test
>
> =D thankz for the fast answer
>
> Hadley Rich escreveu:
>
> On Friday 12 December 2008 09:16:56 Vinicius Kobashi wrote:
>
>
> i found another module called mod_xml_curl and loaded it to freeswitch
> too... but still it shows me the following error:
>
>  2008-12-11 17:04:04 [WARNING] sofia_reg.c:1501 sofia_reg_parse_auth()
> Can't find user [username at freeswitchserver.com] You must define a domain
> called 'freeswitchserver.com' in your directory and add a user with the
> id="username" attribute and you must configure your device to use the
> proper domain in it's authentication credentials.
>
>  does anyone got an idea?
>
>
> Yes, you need to define a domain called 'freeswitchserver.com' in your
> directory and add a user with the id="username" just like the error message
> says.
>
> The directory files are in conf/directory/
>
> If you would like to read up on mod_xml_curl there is a detailed page on the
> wiki;
>
> http://wiki.freeswitch.org/wiki/Mod_xml_curl
>
> hads
>
>
> --
>
>
> Vinicius Kobashi
> Infra-Estrutura
>
> Ydea Desenvolvimento de Software LTDA.
> Av. Adolfo Pinheiro, 2338 - Alto da Boa Vista
> CEP.:04734-004 - S?o Paulo - SP
> Tel.: 55-11-5523-0333
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From erick at junctionnetworks.com  Thu Dec 11 12:52:38 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Thu, 11 Dec 2008 15:52:38 -0500
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
References: 493ED9E6.3000803@junctionnetworks.com
Message-ID: <49417D96.1090805@junctionnetworks.com>

Thanks Dave,

Actually I realized my problem (stupid mistake of course).  For anyone else
trying to use the fs_path variable the value needs to be a fully 
qualified SIP
URI, e.g. "bob at bar.com;fs_path=sip:host.domain.net", notice it being 
prefaced
with the "sip:", my problem was that  I was only entering
the host name.  Then somewhere down in mod_sofia it must have decided that
it didn't like that and just closed the channel.

Hope this helps somebody who gets stuck like I did.

Cheers,

Erick

> Hi Erick,
>
> Not sure if you've tried this (or if it'll help), but you can force
> routing in the dialplan like so:
> <action application="set" data="sip_h_Route=<sip:@11.22.33.44;lr>" />
> <action application="bridge" data="sofia/gateway/gw/$1"/>
>
> Cheers --
>
> Dave




From msc at freeswitch.org  Thu Dec 11 12:59:46 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Dec 2008 12:59:46 -0800
Subject: [Freeswitch-users] Sending SIP calls via outbound proxy
In-Reply-To: <49417D96.1090805@junctionnetworks.com>
References: <49417D96.1090805@junctionnetworks.com>
Message-ID: <87f2f3b90812111259ob0a78bbw747afdc52251c2cb@mail.gmail.com>

On Thu, Dec 11, 2008 at 12:52 PM, Erick Johnson
<erick at junctionnetworks.com> wrote:
> Thanks Dave,
>
> Actually I realized my problem (stupid mistake of course).  For anyone else
> trying to use the fs_path variable the value needs to be a fully
> qualified SIP
> URI, e.g. "bob at bar.com;fs_path=sip:host.domain.net", notice it being
> prefaced
> with the "sip:", my problem was that  I was only entering
> the host name.  Then somewhere down in mod_sofia it must have decided that
> it didn't like that and just closed the channel.

Erick, thanks for the clarification! I'll get it put on the wiki right away.
-MC

>
> Hope this helps somebody who gets stuck like I did.
>
> Cheers,
>
> Erick
>
>> Hi Erick,
>>
>> Not sure if you've tried this (or if it'll help), but you can force
>> routing in the dialplan like so:
>> <action application="set" data="sip_h_Route=<sip:@11.22.33.44;lr>" />
>> <action application="bridge" data="sofia/gateway/gw/$1"/>
>>
>> Cheers --
>>
>> Dave
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From vkobashi at ydeasolutions.com.br  Thu Dec 11 14:49:23 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Thu, 11 Dec 2008 19:49:23 -0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>
	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>
Message-ID: <494198F3.10806@ydeasolutions.com.br>

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From erick at junctionnetworks.com  Thu Dec 11 13:08:52 2008
From: erick at junctionnetworks.com (Erick Johnson)
Date: Thu, 11 Dec 2008 16:08:52 -0500
Subject: [Freeswitch-users]  Restricting SIP methods in Allow header
Message-ID: <49418164.7080206@junctionnetworks.com>

Is it possible it configure a sip-profile so that the UA reports a 
restricted
set of methods in the SIP allow header?  For instance, I would like to 
remove
all methods except  ACK, BYE, CANCEL, NOTIFY, and PRACK

Thanks

Erick

-- 
Erick Johnson



From anthony.minessale at gmail.com  Thu Dec 11 14:36:20 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Dec 2008 16:36:20 -0600
Subject: [Freeswitch-users] Restricting SIP methods in Allow header
In-Reply-To: <49418164.7080206@junctionnetworks.com>
References: <49418164.7080206@junctionnetworks.com>
Message-ID: <191c3a030812111436v2972290bsf889ce2b5db4488@mail.gmail.com>

no but there are many options in the profile that directly control what
methods we advertise based on if they are enabled.


On Thu, Dec 11, 2008 at 3:08 PM, Erick Johnson
<erick at junctionnetworks.com>wrote:

> Is it possible it configure a sip-profile so that the UA reports a
> restricted
> set of methods in the SIP allow header?  For instance, I would like to
> remove
> all methods except  ACK, BYE, CANCEL, NOTIFY, and PRACK
>
> Thanks
>
> Erick
>
> --
> Erick Johnson
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
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From anthony.minessale at gmail.com  Thu Dec 11 15:39:26 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Dec 2008 17:39:26 -0600
Subject: [Freeswitch-users] segmentation fault by auto record
In-Reply-To: <1228978862.31447.5.camel@quentusrex-desktop>
References: <20935513.post@talk.nabble.com>
	<87f2f3b90812100753i664e7c6br12e780fe3be0eacd@mail.gmail.com>
	<8AC8C043-1C07-428A-A531-F44B3DA5BA6F@freeswitch.org>
	<20950299.post@talk.nabble.com>
	<1228978862.31447.5.camel@quentusrex-desktop>
Message-ID: <191c3a030812111539k5b181c96qbfa13f5015c8bb98@mail.gmail.com>

issue should be fixed in SVN r10723

On Thu, Dec 11, 2008 at 1:01 AM, William King <william at channelxstream.com>wrote:

> Carole,
>
> 1.0.1 is known to be broken now. Can you go into the directory where you
> have the freeswitch source from the svn repo and type 'make current'.
> This will update freeswitch to trunk.
>
> -William
>
> On Wed, 2008-12-10 at 22:51 -0800, Carole O. wrote:
> > Hello,
> >
> > I am running the version 1.0.1.
> > Do you still need me to run the debugging?
> >
> > Carole
> >
> >
> > Brian West-3 wrote:
> > >
> > > I have already labbed this up on SVN trunk and I don't get a segfault
> > > but I get something else that prevents it from working properly.    We
> > > are working on it today.  Also what version are you running?
> > >
> > > /b
> > >
> > > On Dec 10, 2008, at 9:53 AM, Michael Collins wrote:
> > >
> > >> Thanks for reporting this. It would be helpful to know a bit more. Can
> > >> you start freeswitch and press F12 (or type "version" at the CLI) and
> > >> report back what it says?
> > >> Also, a backtrace (bt) is generally useful. If you could produce a
> > >> "bt" and a "bt full" from you core file that would be extremely
> > >> helpful.
> > >>
> > >> see this link for more information:
> > >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
> > >>
> > >> you should have a "core" file for each segfault that occurred. Use the
> > >> gdb program to get the back trace:
> > >>
> > >> gdb /path/to/fs/binary core.xxx
> > >>
> > >> then capture the output from these two commands:
> > >>
> > >> bt <enter>
> > >> bt full <enter>
> > >>
> > >> When you type those commands you'll see tons of debugging info;
> > >> capture that and put it in a pastebin (pastebin.freeswitch.org) then
> > >> report back here.
> > >>
> > >> You can exit the gdb debugger by typing q <enter>
> > >>
> > >> Thanks for helping us collect information!
> > >>
> > >> -MC
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> --
> William King
> Cell: 253-686-5518
> E-mail: william at channelxstream.com
> Channel XStream
> www.channelxstream.com
> 1-877-600-6786
>
> If there is a possibility that any information in our conversation might
> be considered 'private' or 'sensitive' such as passwords, account
> information, legal or financial information, or anything else that you
> would consider 'private' or 'sensitive' communications. It is better to
> always err on the side of security. Please encrypt the e-mail using
> my gpg key: 95C9D5B3.
>
> If you are unfamiliar with e-mail encryption feel free to let me know
> and I can help you establish the proper protocols and procedures.
> https://help.ubuntu.com/community/GnuPrivacyGuardHowto
>
> Get my gpg key: gpg --recv-key --keyserver keyserver.ubuntu.com 95C9D5B3
> Key Fingerprint: EA6F B2EE 1846 55D4 FFD9  80BA 6489 B48C 95C9 D5B3
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
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FreeSWITCH Developer Conference
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From dalechase at yahoo.com  Thu Dec 11 16:51:30 2008
From: dalechase at yahoo.com (dalechase at yahoo.com)
Date: Thu, 11 Dec 2008 16:51:30 -0800 (PST)
Subject: [Freeswitch-users] config help: openzap and T1 A102u
Message-ID: <603978.74977.qm@web36108.mail.mud.yahoo.com>

I am stuck trying to bring up freeswitch with openzap on a Sangoma A102u T1 card.
Works fine with asterisk.

Please point out where I am failing to configure properly.

Running Linux version 2.6.9-34.ELsmp on a Dell Celeron

% wanrouter hwprobe verbose

-----------------------------------------
| Wanpipe Hardware Probe Info (verbose) |
-----------------------------------------
1 . AFT-A102u  : SLOT=1 : BUS=2 : IRQ=145 : CPU=A : PORT=PRI : V=25
+01:PMC4351:PCI
2 . AFT-A102u  : SLOT=1 : BUS=2 : IRQ=145 : CPU=B : PORT=PRI : V=25
+01:PMC4351:PCI

Card Cnt: S508=0  S514X=0  S518=0  A101-2=1  A104=0  A300=0  A200=0  A108=0

% cat /usr/local/freeswitch/conf/autoload_configs/open
openmrcp.conf.xml  openzap.conf.xml
[root at pbxtra1466 freeswitch]# cat /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
  <!-- one entry here per openzap span -->
  <analog_spans>
    <span id="1">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <!--<param name="enable-analog-option" value="call-swap"/>-->
      <!--<param name="enable-analog-option" value="3-way"/>-->
      <param name="tonegroup" value="us"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
    <span id="2">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <!--<param name="enable-analog-option" value="call-swap"/>-->
      <!--<param name="enable-analog-option" value="3-way"/>-->
      <param name="tonegroup" value="us"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </analog_spans>

</configuration>

% cat /etc/openzap/openzap.conf
[span wanpipe]
trunk_type => t1
b-channel => 1:1-23
d-channel=> 1:24

[span wanpipe]
trunk_type => t1
b-channel => 2:25-47
d-channel=> 2:48

% cat /etc/openzap/wanpipe.conf
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750

% cat /etc/wanpipe/wanpipe1.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Tue Dec 12 16:21:45 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/sbin/wancfg program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe1 = WAN_AFT, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE       = AFT
S514CPU         = A
CommPort        = PRI
AUTO_PCISLOT    = NO
PCISLOT         = 1
PCIBUS          = 2
FE_MEDIA        = T1
FE_LCODE        = B8ZS
FE_FRAME        = ESF
FE_LINE         = 1
TE_CLOCK        = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE        = NO
LBO             = 0DB
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END = NO
TDMV_SPAN       = 1
TDMV_DCHAN      = 24

[w1g1]
ACTIVE_CH       = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC       = NO

% cat /etc/wanpipe/wanpipe2.conf
#================================================
# WANPIPE1 Configuration File
#================================================
#
# Date: Tue Dec 12 16:21:45 UTC 2006
#
# Note: This file was generated automatically
#       by /usr/sbin/wancfg program.
#
#       If you want to edit this file, it is
#       recommended that you use wancfg program
#       to do so.
#================================================
# Sangoma Technologies Inc.
#================================================

[devices]
wanpipe2 = WAN_AFT, Comment

[interfaces]
w2g1 = wanpipe2, , TDM_VOICE, Comment

[wanpipe2]
CARD_TYPE       = AFT
S514CPU         = B
CommPort        = PRI
AUTO_PCISLOT    = NO
PCISLOT         = 1
PCIBUS          = 2
FE_MEDIA        = T1
FE_LCODE        = B8ZS
FE_FRAME        = ESF
FE_LINE         = 1
TE_CLOCK        = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE     = CCS
TE_HIGHIMPEDANCE        = NO
LBO             = 0DB
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END = NO
TDMV_SPAN       = 2
TDMV_DCHAN      = 24

[w2g1]
ACTIVE_CH       = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC       = NO

freeswitch at hostname-elided> load mod_openzap
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c1
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c2
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c3
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c4
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c5
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c6
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c7
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c8
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c9
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c10
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c11
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c12
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c13
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c14
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c15
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c16
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c17
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c18
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c19
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c20
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c21
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c22
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c23
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s1c24
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c25
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c26
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c27
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c28
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c29
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c30
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c31
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c32
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c33
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c34
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c35
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c36
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c37
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c38
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c39
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c40
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c41
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c42
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c43
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c44
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c45
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c46
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c47
2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure configuring device s2c48
2008-12-11 16:23:08 [INFO] zap_io.c:2068 load_config() Configured 0 channel(s)
2008-12-11 16:23:08 [ERR] zap_io.c:2161 zap_global_init() No modules configured!
2008-12-11 16:23:08 [ERR] mod_openzap.c:1861 mod_openzap_load() Error loading OpenZAP
2008-12-11 16:23:08 [CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so
**Module load routine returned an error**


      



From zolotov at altron.ua  Fri Dec 12 01:00:01 2008
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Fri, 12 Dec 2008 11:00:01 +0200
Subject: [Freeswitch-users] config help: openzap and T1 A102u
References: <603978.74977.qm@web36108.mail.mud.yahoo.com>
Message-ID: <004001c95c38$04da0210$6d02a8c0@opos20>

  Did you try ./wanrouter start before starting FreeSWITCH ?
----- Original Message ----- 
From: <dalechase at yahoo.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, December 12, 2008 2:51 AM
Subject: [Freeswitch-users] config help: openzap and T1 A102u


>I am stuck trying to bring up freeswitch with openzap on a Sangoma A102u T1 
>card.
> Works fine with asterisk.
>
> Please point out where I am failing to configure properly.
>
> Running Linux version 2.6.9-34.ELsmp on a Dell Celeron
>
> % wanrouter hwprobe verbose
>
> -----------------------------------------
> | Wanpipe Hardware Probe Info (verbose) |
> -----------------------------------------
> 1 . AFT-A102u  : SLOT=1 : BUS=2 : IRQ=145 : CPU=A : PORT=PRI : V=25
> +01:PMC4351:PCI
> 2 . AFT-A102u  : SLOT=1 : BUS=2 : IRQ=145 : CPU=B : PORT=PRI : V=25
> +01:PMC4351:PCI
>
> Card Cnt: S508=0  S514X=0  S518=0  A101-2=1  A104=0  A300=0  A200=0 
> A108=0
>
> % cat /usr/local/freeswitch/conf/autoload_configs/open
> openmrcp.conf.xml  openzap.conf.xml
> [root at pbxtra1466 freeswitch]# cat 
> /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="0"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>  <!-- one entry here per openzap span -->
>  <analog_spans>
>    <span id="1">
>      <!--<param name="hold-music" value="$${moh_uri}"/>-->
>      <!--<param name="enable-analog-option" value="call-swap"/>-->
>      <!--<param name="enable-analog-option" value="3-way"/>-->
>      <param name="tonegroup" value="us"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>      <!-- regex to stop dialing when it matches -->
>      <!--<param name="dial-regex" value="5555"/>-->
>      <!-- regex to stop dialing when it does not match -->
>      <!--<param name="fail-dial-regex" value="^5"/>-->
>    </span>
>    <span id="2">
>      <!--<param name="hold-music" value="$${moh_uri}"/>-->
>      <!--<param name="enable-analog-option" value="call-swap"/>-->
>      <!--<param name="enable-analog-option" value="3-way"/>-->
>      <param name="tonegroup" value="us"/>
>      <param name="digit-timeout" value="2000"/>
>      <param name="max-digits" value="11"/>
>      <param name="dialplan" value="XML"/>
>      <param name="context" value="default"/>
>      <!-- regex to stop dialing when it matches -->
>      <!--<param name="dial-regex" value="5555"/>-->
>      <!-- regex to stop dialing when it does not match -->
>      <!--<param name="fail-dial-regex" value="^5"/>-->
>    </span>
>  </analog_spans>
>
> </configuration>
>
> % cat /etc/openzap/openzap.conf
> [span wanpipe]
> trunk_type => t1
> b-channel => 1:1-23
> d-channel=> 1:24
>
> [span wanpipe]
> trunk_type => t1
> b-channel => 2:25-47
> d-channel=> 2:48
>
> % cat /etc/openzap/wanpipe.conf
> [defaults]
> codec_ms => 20
> wink_ms => 150
> flash_ms => 750
>
> % cat /etc/wanpipe/wanpipe1.conf
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Date: Tue Dec 12 16:21:45 UTC 2006
> #
> # Note: This file was generated automatically
> #       by /usr/sbin/wancfg program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
> [devices]
> wanpipe1 = WAN_AFT, Comment
>
> [interfaces]
> w1g1 = wanpipe1, , TDM_VOICE, Comment
>
> [wanpipe1]
> CARD_TYPE       = AFT
> S514CPU         = A
> CommPort        = PRI
> AUTO_PCISLOT    = NO
> PCISLOT         = 1
> PCIBUS          = 2
> FE_MEDIA        = T1
> FE_LCODE        = B8ZS
> FE_FRAME        = ESF
> FE_LINE         = 1
> TE_CLOCK        = NORMAL
> TE_REF_CLOCK    = 0
> TE_SIG_MODE     = CCS
> TE_HIGHIMPEDANCE        = NO
> LBO             = 0DB
> FE_TXTRISTATE   = NO
> MTU             = 1500
> UDPPORT         = 9000
> TTL             = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN       = 1
> TDMV_DCHAN      = 24
>
> [w1g1]
> ACTIVE_CH       = ALL
> TDMV_ECHO_OFF   = NO
> TDMV_HWEC       = NO
>
> % cat /etc/wanpipe/wanpipe2.conf
> #================================================
> # WANPIPE1 Configuration File
> #================================================
> #
> # Date: Tue Dec 12 16:21:45 UTC 2006
> #
> # Note: This file was generated automatically
> #       by /usr/sbin/wancfg program.
> #
> #       If you want to edit this file, it is
> #       recommended that you use wancfg program
> #       to do so.
> #================================================
> # Sangoma Technologies Inc.
> #================================================
>
> [devices]
> wanpipe2 = WAN_AFT, Comment
>
> [interfaces]
> w2g1 = wanpipe2, , TDM_VOICE, Comment
>
> [wanpipe2]
> CARD_TYPE       = AFT
> S514CPU         = B
> CommPort        = PRI
> AUTO_PCISLOT    = NO
> PCISLOT         = 1
> PCIBUS          = 2
> FE_MEDIA        = T1
> FE_LCODE        = B8ZS
> FE_FRAME        = ESF
> FE_LINE         = 1
> TE_CLOCK        = NORMAL
> TE_REF_CLOCK    = 0
> TE_SIG_MODE     = CCS
> TE_HIGHIMPEDANCE        = NO
> LBO             = 0DB
> FE_TXTRISTATE   = NO
> MTU             = 1500
> UDPPORT         = 9000
> TTL             = 255
> IGNORE_FRONT_END = NO
> TDMV_SPAN       = 2
> TDMV_DCHAN      = 24
>
> [w2g1]
> ACTIVE_CH       = ALL
> TDMV_ECHO_OFF   = NO
> TDMV_HWEC       = NO
>
> freeswitch at hostname-elided> load mod_openzap
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c1
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c2
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c3
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c4
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c5
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c6
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c7
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c8
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c9
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c10
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c11
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c12
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c13
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c14
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c15
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c16
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c17
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c18
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c19
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c20
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c21
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c22
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c23
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s1c24
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c25
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c26
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c27
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c28
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c29
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c30
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c31
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c32
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c33
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c34
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c35
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c36
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c37
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c38
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c39
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c40
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c41
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c42
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c43
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c44
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c45
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c46
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c47
> 2008-12-11 16:23:08 [ERR] zap_wanpipe.c:414 wp_open_range() failure 
> configuring device s2c48
> 2008-12-11 16:23:08 [INFO] zap_io.c:2068 load_config() Configured 0 
> channel(s)
> 2008-12-11 16:23:08 [ERR] zap_io.c:2161 zap_global_init() No modules 
> configured!
> 2008-12-11 16:23:08 [ERR] mod_openzap.c:1861 mod_openzap_load() Error 
> loading OpenZAP
> 2008-12-11 16:23:08 [CRIT] switch_loadable_module.c:756 
> switch_loadable_module_load_file() Error Loading module 
> /usr/local/freeswitch/mod/mod_openzap.so
> **Module load routine returned an error**
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 




From alex at sinapticode.ro  Fri Dec 12 03:38:47 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 13:38:47 +0200
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers
Message-ID: <1229081927.4100.7.camel@gathern.lan>

Hi,

I'm working on a simple dialer, and I have the following problem: the
audio file starts playing before the user answeres the phone (while it's
ringing). It only works when I introduce a delay, but that doesn't seem
right.

For instance in the asterisk context referred in the call files, I had:

exten => s,4,Answer
exten => s,n,Wait(2)
exten => s,n,Background(${SOUNDFILE})
And indeed it played a soundfile 2 seconds after the called person
picked up the phone

In FS I currently initiate calls like this:

session.waitForAnswer(10000);

if (session.ready()) {
    session.sleep(2000);
    session.streamFile(/*...*/);
}

Is this right?




From alex at sinapticode.ro  Fri Dec 12 04:10:29 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 14:10:29 +0200
Subject: [Freeswitch-users] Freeswitch logging
Message-ID: <1229083829.4100.11.camel@gathern.lan>

Hi,

I see that mod_cdr is marked as being non-functional on the wiki. I'm working on a dialer and I need a way to log information about calls.

What module should I use?

Thanks,




From hads at nice.net.nz  Fri Dec 12 04:26:56 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Sat, 13 Dec 2008 01:26:56 +1300
Subject: [Freeswitch-users] Freeswitch logging
In-Reply-To: <1229083829.4100.11.camel@gathern.lan>
References: <1229083829.4100.11.camel@gathern.lan>
Message-ID: <200812130126.56386.hads@nice.net.nz>

On Saturday 13 December 2008 01:10:29 Alexandru Nedelcu wrote:
> Hi,
>
> I see that mod_cdr is marked as being non-functional on the wiki. I'm
> working on a dialer and I need a way to log information about calls.
>
> What module should I use?

This was answered on IRC and a note added to the mod_cdr wiki page.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier



From jbr at consiglia.dk  Fri Dec 12 05:30:27 2008
From: jbr at consiglia.dk (Jon Bruel)
Date: Fri, 12 Dec 2008 14:30:27 +0100
Subject: [Freeswitch-users] fifo.conf.xml usage
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE027E3B3E6E@SBS2008SERVER.consiglia.local>

I'm happy to see that you can add consumers to queues using the fifo.conf.xml configuration file. I have made some tests and I hope it may lead to a more universal way of setting up queues for small organisations than the one I have described in the wiki, and which includes (too) many javascripts. I have some questions to clarify my understanding. Using the fifo.conf.xml, I find:
 - That the consumers continue to ring after the caller has abandoned the queue. Is there a way to avoid this?
Further:
 - Is there a way to control the caller_id_name/number presented to the consumer?
 - Is there a way to control the ringing tone in the consumers like the one which can be used in the dialplan?
 - Can the fifo.conf.xml refer to an ODBC connection in order to get the members from a database?
Finally, thanks for all the good work everybody in the FS community has put into FS, I truly believe in the possibilities of this product. Checking the hits on Google certainly indicates you moving into the right direction. /Jon

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From d at d-man.org  Fri Dec 12 05:48:14 2008
From: d at d-man.org (Darren Schreiber)
Date: Fri, 12 Dec 2008 05:48:14 -0800
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers
In-Reply-To: <1229081927.4100.7.camel@gathern.lan>
References: <1229081927.4100.7.camel@gathern.lan>
Message-ID: <05FEA4243A6C422DB5A3D7838AA709FE@test>

How are you originating calls? You probably need to add
{ignore_early_media=true}. This tells FreeSWITCH not to return from
origination when early media (progress/ringing) was received (I think
anyway)...

See http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media

There is a sample of this in use with the originate command here:
http://wiki.freeswitch.org/wiki/Mod_commands#originate (about halfway down)

Setting channel variables before doing the originate 

   originate {ignore_early_media=true}sofia/mydomain.com/18005551212 at 1.2.3.4
15555551212



Since you are making a dialer, you may want to start the originations in the
background and move on to the next call while tweaking the timeout value for
originated calls. From the WIKI again:

"You can originate a call in the background (asynchronously) and playback a
message with a 60 second timeout. 

   bgapi originate
{ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number
&playback(message)"

- Darren

 

-----Original Message-----
From: Alexandru Nedelcu [mailto:alex at sinapticode.ro] 
Sent: Friday, December 12, 2008 3:39 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers

Hi,

I'm working on a simple dialer, and I have the following problem: the audio
file starts playing before the user answeres the phone (while it's ringing).
It only works when I introduce a delay, but that doesn't seem right.

For instance in the asterisk context referred in the call files, I had:

exten => s,4,Answer
exten => s,n,Wait(2)
exten => s,n,Background(${SOUNDFILE})
And indeed it played a soundfile 2 seconds after the called person picked up
the phone

In FS I currently initiate calls like this:

session.waitForAnswer(10000);

if (session.ready()) {
    session.sleep(2000);
    session.streamFile(/*...*/);
}

Is this right?


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From carole.olivier at enst.fr  Fri Dec 12 06:26:27 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Fri, 12 Dec 2008 06:26:27 -0800 (PST)
Subject: [Freeswitch-users] conference_auto_outcall_announce
In-Reply-To: <20956587.post@talk.nabble.com>
References: <20955216.post@talk.nabble.com>
	<0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>
	<20956587.post@talk.nabble.com>
Message-ID: <20976612.post@talk.nabble.com>


Hello,

First, I would like to apologize for a mistake I have made: by adding the
following line in the profile
< param name="enter-sound" value="path/to/file.wav" / >
the enter sound is played.
I am sorry for this. I did not hear it because in the case I have been
analyzing the members of the conference the caller automatically invites are
VoIP speakers which beep before playing anything and apparently miss the
enter sound. (both the beep and the enter-sound have about the same length).

I still have the following questions:
1- Is it possible to introduce a delay so that the enter sound is played
only after 2s?

2- I have noticed that if the caller of the conference talks or makes some
noises at the very beginning when he is entering the conference and the
enter sound is played, we can hear it through the VoIP speakers. Is there
any way to prevent from this? I would like to mute the caller during the
enter-sound and I would need this to be done statically, I mean in the xml
files, and not from the shell.

Thanks!!
Carole



Carole O. wrote:
> 
> Hello,
> 
> Actually, I have already tried it but nothing happens: the file is not
> played and there is no error. 
> There is still a difference: if I configure it as you said, I can not be
> listening anymore, there is simply nothing.
> 
> Would you have an idea? I have checked the path and the syntax 1 million
> times so I do not think I make mistake there.
> 
> Thanks,
> Carole
> 
> 
> 
> Brian West-3 wrote:
>> 
>> Don't have play: in there and it should be fine.  Also if you want the  
>> absolute path you start it with /path/to/file.wav
>> 
>> 
>> /b
>> 
>> On Dec 11, 2008, at 7:13 AM, Carole O. wrote:
>> 
>>> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
>>> [System error : no such file or directory]
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 

-- 
View this message in context: http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20976612.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Fri Dec 12 06:54:42 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 08:54:42 -0600
Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
In-Reply-To: <8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
References: <DD63B9EA-92F4-4E32-94C3-26D80B2E366A@freeswitch.org>
	<8CB291E3E805A95-9D4-66C@Webmail-mg06.sim.aol.com>
Message-ID: <8329151B-91CE-4F18-99C3-1567DD82D210@freeswitch.org>

FreeSWITCHers,
	I would like to thank everyone that donated.  Enough was raised to  
cover the sound order.  ;)

Thanks,
Brian West
FreeSWITCH.org




From helmut.kuper at ewetel.de  Fri Dec 12 07:01:26 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 12 Dec 2008 16:01:26 +0100
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
Message-ID: <49427CC6.2090407@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
server in a non-root environment.

We experienced a timer problem which led to this FS console error message:

[ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries


During anylizing this we found that q921 T203 is never reset when link
is in state "Multiple Frame Mode Established" and SABME frames are
received by FS. So it must timeout regardless if SABME frames are
received or not.
Additionally we found that the default T203 value (10 sec) was too short
for AVAYA (it has to be >=19 sec)

To fix the problem we changed two things in q921.c:

Change T203 default value from 10 sec to 20000 sec
Line 406: trunk->T203Timeout = 20000;

Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
received SABME frame
Line 1996: Q921T203TimerReset(trunk, tei);

After recompiling FS the Error disapeared. Next week we will do some
calls over the link to make sure there are no other side effects.

Is it planned to make the q921 timeouts configurable in openzap.conf or
in openzap.conf.xml?

best regards
Helmut


PS: My openzap configs:

openzap.conf

[span wanpipe PRI_1]
trunk_type => E1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31




openzap.conf.xml

<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="7"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
   <pri_spans>
     <span name="PRI_1">
       <!-- Log Levels: none, alert, crit, err, warning, notice, info,
debug -->
       <param name="q921loglevel" value="debug"/>
       <param name="q931loglevel" value="debug"/>
       <param name="mode" value="user"/>
       <param name="dialect" value="Oh this is not my dialect"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="default"/>
     </span>
   </pri_spans>
</configuration>

Very interesting here is, that the "dialect" parameter doesn't seem to
have an effect on FS. I use that one above without any errors or warning
and I guess that was not intended.



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From vkobashi at ydeasolutions.com.br  Fri Dec 12 08:15:33 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Fri, 12 Dec 2008 13:15:33 -0300
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <494198F3.10806@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>
	<494198F3.10806@ydeasolutions.com.br>
Message-ID: <49428E25.80209@ydeasolutions.com.br>

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From anthony.minessale at gmail.com  Fri Dec 12 07:15:25 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2008 09:15:25 -0600
Subject: [Freeswitch-users] fifo.conf.xml usage
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE027E3B3E6E@SBS2008SERVER.consiglia.local>
References: <AclcXcuC05uCObneTKeV2P/QtO/NXQ==>
	<DE9AFF5DF0C8BB4881F86E50EAC0A9AE027E3B3E6E@SBS2008SERVER.consiglia.local>
Message-ID: <191c3a030812120715y5ad0b0e9i639006a32d72afea@mail.gmail.com>

the entries are standard originate strings so all of the {} variables apply.


On Fri, Dec 12, 2008 at 7:30 AM, Jon Bruel <jbr at consiglia.dk> wrote:

>  I'm happy to see that you can add consumers to queues using the
> fifo.conf.xml configuration file. I have made some tests and I hope it may
> lead to a more universal way of setting up queues for small organisations
> than the one I have described in the wiki, and which includes (too) many
> javascripts. I have some questions to clarify my understanding. Using the
> fifo.conf.xml, I find:
>
>  - That the consumers continue to ring after the caller has abandoned the
> queue. Is there a way to avoid this?
>
> Further:
>
>  - Is there a way to control the caller_id_name/number presented to the
> consumer?
>
>  - Is there a way to control the ringing tone in the consumers like the one
> which can be used in the dialplan?
>
>  - Can the fifo.conf.xml refer to an ODBC connection in order to get the
> members from a database?
>
> Finally, thanks for all the good work everybody in the FS community has put
> into FS, I truly believe in the possibilities of this product. Checking the
> hits on Google certainly indicates you moving into the right direction. /Jon
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Fri Dec 12 07:18:09 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2008 09:18:09 -0600
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <49427CC6.2090407@ewetel.de>
References: <49427CC6.2090407@ewetel.de>
Message-ID: <191c3a030812120718n7d8c5410y2ad3cebab8f5be3b@mail.gmail.com>

if you open a jira issue on it we can probably add your patch and/or the
config option.
the users-list is a tough place to manage TDM issues.


On Fri, Dec 12, 2008 at 9:01 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
> AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
> server in a non-root environment.
>
> We experienced a timer problem which led to this FS console error message:
>
> [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries
>
>
> During anylizing this we found that q921 T203 is never reset when link
> is in state "Multiple Frame Mode Established" and SABME frames are
> received by FS. So it must timeout regardless if SABME frames are
> received or not.
> Additionally we found that the default T203 value (10 sec) was too short
> for AVAYA (it has to be >=19 sec)
>
> To fix the problem we changed two things in q921.c:
>
> Change T203 default value from 10 sec to 20000 sec
> Line 406: trunk->T203Timeout = 20000;
>
> Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
> received SABME frame
> Line 1996: Q921T203TimerReset(trunk, tei);
>
> After recompiling FS the Error disapeared. Next week we will do some
> calls over the link to make sure there are no other side effects.
>
> Is it planned to make the q921 timeouts configurable in openzap.conf or
> in openzap.conf.xml?
>
> best regards
> Helmut
>
>
> PS: My openzap configs:
>
> openzap.conf
>
> [span wanpipe PRI_1]
> trunk_type => E1
> b-channel => 1:1-15
> d-channel => 1:16
> b-channel => 1:17-31
>
>
>
>
> openzap.conf.xml
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="7"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>   <pri_spans>
>     <span name="PRI_1">
>       <!-- Log Levels: none, alert, crit, err, warning, notice, info,
> debug -->
>       <param name="q921loglevel" value="debug"/>
>       <param name="q931loglevel" value="debug"/>
>       <param name="mode" value="user"/>
>       <param name="dialect" value="Oh this is not my dialect"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>     </span>
>   </pri_spans>
> </configuration>
>
> Very interesting here is, that the "dialect" parameter doesn't seem to
> have an effect on FS. I use that one above without any errors or warning
> and I guess that was not intended.
>
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
> BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
> =geXp
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Dec 12 07:22:53 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2008 09:22:53 -0600
Subject: [Freeswitch-users] conference_auto_outcall_announce
In-Reply-To: <20976612.post@talk.nabble.com>
References: <20955216.post@talk.nabble.com>
	<0DB9905B-4A03-4F24-B0A8-BB23ACE3567D@freeswitch.org>
	<20956587.post@talk.nabble.com> <20976612.post@talk.nabble.com>
Message-ID: <191c3a030812120722y7749c160x28e13474ed878943@mail.gmail.com>

No,
there is currently no way.


On Fri, Dec 12, 2008 at 8:26 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> Hello,
>
> First, I would like to apologize for a mistake I have made: by adding the
> following line in the profile
> < param name="enter-sound" value="path/to/file.wav" / >
> the enter sound is played.
> I am sorry for this. I did not hear it because in the case I have been
> analyzing the members of the conference the caller automatically invites
> are
> VoIP speakers which beep before playing anything and apparently miss the
> enter sound. (both the beep and the enter-sound have about the same
> length).
>
> I still have the following questions:
> 1- Is it possible to introduce a delay so that the enter sound is played
> only after 2s?
>
> 2- I have noticed that if the caller of the conference talks or makes some
> noises at the very beginning when he is entering the conference and the
> enter sound is played, we can hear it through the VoIP speakers. Is there
> any way to prevent from this? I would like to mute the caller during the
> enter-sound and I would need this to be done statically, I mean in the xml
> files, and not from the shell.
>
> Thanks!!
> Carole
>
>
>
> Carole O. wrote:
> >
> > Hello,
> >
> > Actually, I have already tried it but nothing happens: the file is not
> > played and there is no error.
> > There is still a difference: if I configure it as you said, I can not be
> > listening anymore, there is simply nothing.
> >
> > Would you have an idea? I have checked the path and the syntax 1 million
> > times so I do not think I make mistake there.
> >
> > Thanks,
> > Carole
> >
> >
> >
> > Brian West-3 wrote:
> >>
> >> Don't have play: in there and it should be fine.  Also if you want the
> >> absolute path you start it with /path/to/file.wav
> >>
> >>
> >> /b
> >>
> >> On Dec 11, 2008, at 7:13 AM, Carole O. wrote:
> >>
> >>> [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
> >>> [System error : no such file or directory]
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20976612.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jaugenstine at gmail.com  Fri Dec 12 09:08:41 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Fri, 12 Dec 2008 09:08:41 -0800
Subject: [Freeswitch-users]  call transfer question
Message-ID: <207e7a5e0812120908w1307ee18j5288015132ed3f3e@mail.gmail.com>

I have a call scenario that involves transferring the call and dropping out
of the SIP/RTP stream.  I need to accept the SIP call, play a prompt, and
retrieve a pin code.  After a database lookup, I need to transfer the call
to another FS server and drop out of the SIP path.  I have done this with
the RTP media stream previously.  I am not sure what I need to do to drop
out of the SIP path.  Is this possible on FS?

Jonathan
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From brian at freeswitch.org  Fri Dec 12 09:14:59 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 11:14:59 -0600
Subject: [Freeswitch-users] call transfer question
In-Reply-To: <207e7a5e0812120908w1307ee18j5288015132ed3f3e@mail.gmail.com>
References: <207e7a5e0812120908w1307ee18j5288015132ed3f3e@mail.gmail.com>
Message-ID: <F4FB83DA-79C2-4E68-928A-223E2883B74C@freeswitch.org>

You can use deflect to accomplish this.. it will do a refer to the  
other FS box.

/b

On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote:

> I have a call scenario that involves transferring the call and  
> dropping out of the SIP/RTP stream.  I need to accept the SIP call,  
> play a prompt, and retrieve a pin code.  After a database lookup, I  
> need to transfer the call to another FS server and drop out of the  
> SIP path.  I have done this with the RTP media stream previously.  I  
> am not sure what I need to do to drop out of the SIP path.  Is this  
> possible on FS?
>
> Jonathan




From jaugenstine at gmail.com  Fri Dec 12 09:36:30 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Fri, 12 Dec 2008 09:36:30 -0800
Subject: [Freeswitch-users] call transfer question
In-Reply-To: <F4FB83DA-79C2-4E68-928A-223E2883B74C@freeswitch.org>
References: <207e7a5e0812120908w1307ee18j5288015132ed3f3e@mail.gmail.com>
	<F4FB83DA-79C2-4E68-928A-223E2883B74C@freeswitch.org>
Message-ID: <207e7a5e0812120936m35ac7554mc498d85c003fe282@mail.gmail.com>

Thank you, that is exactly what I need.

On Fri, Dec 12, 2008 at 9:14 AM, Brian West <brian at freeswitch.org> wrote:

> You can use deflect to accomplish this.. it will do a refer to the
> other FS box.
>
> /b
>
> On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote:
>
> > I have a call scenario that involves transferring the call and
> > dropping out of the SIP/RTP stream.  I need to accept the SIP call,
> > play a prompt, and retrieve a pin code.  After a database lookup, I
> > need to transfer the call to another FS server and drop out of the
> > SIP path.  I have done this with the RTP media stream previously.  I
> > am not sure what I need to do to drop out of the SIP path.  Is this
> > possible on FS?
> >
> > Jonathan
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Fri Dec 12 01:38:59 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 12 Dec 2008 20:38:59 +1100
Subject: [Freeswitch-users] Error loading portaudio module
Message-ID: <20081212093859.GA7067@jdc.jasonjgw.net>

I am new to FreeSWITCH; hence this is the first of what will probably be a
number of questions as I learn.

I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
2.6.27, x86_64 architecture), with the portaudio19-dev package installed.

Whenever I try to load the portaudio module I get the following in the logs. I
haven't changed anything in the default portaudio configuration that comes
with FreeSWITCH.

PortAudio version number = 1899
PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
Number of devices = 0
2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
input
 device!
2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
input
 device!
2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
switch_loadable_module_l
oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so

Other software that uses portaudio is known to work. I would expect FreeSWITCH
to detect my Alsa sound devices.

Suggestions welcome.




From alex at sinapticode.ro  Fri Dec 12 10:37:26 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 20:37:26 +0200
Subject: [Freeswitch-users] CDR logs - adding a custom field
Message-ID: <1229107046.4100.18.camel@gathern.lan>

In Asterisk I was able to set a custom CDR field by doing something
like: 
   Set(CDR(userfield)=${SOMETHING})

I need to set a custom field in FreeSwitch, and preferably I want to
have control over its value from Javascript.

Can someone tell me how? :)

Thanks,

-- 
Alexandru Nedelcu
Software Developer, Sinapticode




From anthony.minessale at gmail.com  Fri Dec 12 11:18:32 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2008 13:18:32 -0600
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <1229107046.4100.18.camel@gathern.lan>
References: <1229107046.4100.18.camel@gathern.lan>
Message-ID: <191c3a030812121118y3c339dc1q6697ff8905576c94@mail.gmail.com>

Yes, I'm familiar with that since i invented that feature for Asterisk =D


In FreeSWITCH, All variables are already available from the cdr
just set regular channel variables.

for xml cdr they are all there right away
for csv cdr you can reference any channel variable in your template.




On Fri, Dec 12, 2008 at 12:37 PM, Alexandru Nedelcu <alex at sinapticode.ro>wrote:

> In Asterisk I was able to set a custom CDR field by doing something
> like:
>   Set(CDR(userfield)=${SOMETHING})
>
> I need to set a custom field in FreeSwitch, and preferably I want to
> have control over its value from Javascript.
>
> Can someone tell me how? :)
>
> Thanks,
>
> --
> Alexandru Nedelcu
> Software Developer, Sinapticode
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From msc at freeswitch.org  Fri Dec 12 11:50:10 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 11:50:10 -0800
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <1229107046.4100.18.camel@gathern.lan>
References: <1229107046.4100.18.camel@gathern.lan>
Message-ID: <87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>

Are you using CSV or XML? The reason I ask is because I personally use
XML and I find that having lots of information (even too much) is
better than not enough. The only drawback to XML that I find is that
you have to know how to parse it properly. :) The level of detail in
the XML CDRs is unmatched by any telephony system I've ever
encountered. I highly recommend it.

Also, check out this wiki page if you haven't already:
http://wiki.freeswitch.org/wiki/Mod_xml_cdr

-MC

On Fri, Dec 12, 2008 at 10:37 AM, Alexandru Nedelcu <alex at sinapticode.ro> wrote:
> In Asterisk I was able to set a custom CDR field by doing something
> like:
>   Set(CDR(userfield)=${SOMETHING})
>
> I need to set a custom field in FreeSwitch, and preferably I want to
> have control over its value from Javascript.
>
> Can someone tell me how? :)
>
> Thanks,
>
> --
> Alexandru Nedelcu
> Software Developer, Sinapticode
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From msc at freeswitch.org  Fri Dec 12 11:58:13 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 11:58:13 -0800
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081212093859.GA7067@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>

Jason,

If I understand correctly software other than PA can lock up the sound
card so that PA doesn't "see" it. That might explain why PA reports
number of devices = 0. Could you check to see if possibly something
else has control of your sound card, perhaps ALSA? Turn off anything
that might use the sound card and then restart FS to see if PA can
then detect your device.

-MC

On Fri, Dec 12, 2008 at 1:38 AM, Jason White <jason at jasonjgw.net> wrote:
> I am new to FreeSWITCH; hence this is the first of what will probably be a
> number of questions as I learn.
>
> I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
> 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.
>
> Whenever I try to load the portaudio module I get the following in the logs. I
> haven't changed anything in the default portaudio configuration that comes
> with FreeSWITCH.
>
> PortAudio version number = 1899
> PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
> Number of devices = 0
> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
> input
>  device!
> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
> input
>  device!
> 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
> switch_loadable_module_l
> oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so
>
> Other software that uses portaudio is known to work. I would expect FreeSWITCH
> to detect my Alsa sound devices.
>
> Suggestions welcome.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From alex at sinapticode.ro  Fri Dec 12 12:12:21 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 22:12:21 +0200
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <191c3a030812121118y3c339dc1q6697ff8905576c94@mail.gmail.com>
References: <1229107046.4100.18.camel@gathern.lan>
	<191c3a030812121118y3c339dc1q6697ff8905576c94@mail.gmail.com>
Message-ID: <1229112741.4100.20.camel@gathern.lan>

On Fri, 2008-12-12 at 13:18 -0600, Anthony Minessale wrote:
> Yes, I'm familiar with that since i invented that feature for Asterisk
> =D
> 
> 
> In FreeSWITCH, All variables are already available from the cdr
> just set regular channel variables.
> 
> for xml cdr they are all there right away
> for csv cdr you can reference any channel variable in your template.
> 

Thank you Anthony,

In case someone wants to know how to set channel variables, there's a
link on the wiki here: http://wiki.freeswitch.org/wiki/Channel_Variables





From alex at sinapticode.ro  Fri Dec 12 12:14:33 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 22:14:33 +0200
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
Message-ID: <1229112873.4100.23.camel@gathern.lan>

Thanks Michael,

I'm going to use XML, since I don't really know what variables I want.
Another problem with CSV is that many people parse them with regular
expressions and scripts break when you add a new column.


On Fri, 2008-12-12 at 11:50 -0800, Michael Collins wrote:
> Are you using CSV or XML? The reason I ask is because I personally use
> XML and I find that having lots of information (even too much) is
> better than not enough. The only drawback to XML that I find is that
> you have to know how to parse it properly. :) The level of detail in
> the XML CDRs is unmatched by any telephony system I've ever
> encountered. I highly recommend it.




From alex at sinapticode.ro  Fri Dec 12 12:17:44 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 22:17:44 +0200
Subject: [Freeswitch-users] Freeswitch logging
In-Reply-To: <200812130126.56386.hads@nice.net.nz>
References: <1229083829.4100.11.camel@gathern.lan>
	<200812130126.56386.hads@nice.net.nz>
Message-ID: <1229113064.4100.26.camel@gathern.lan>

On Sat, 2008-12-13 at 01:26 +1300, Hadley Rich wrote:
> This was answered on IRC and a note added to the mod_cdr wiki page.

Thanks Hadley,

I'm a total newbie to FreeSwitch and voip in general, sorry for my
persistence :) I'll try writing an article about my setup this weekend.





From brian at freeswitch.org  Fri Dec 12 12:21:56 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 14:21:56 -0600
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <1229112873.4100.23.camel@gathern.lan>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
Message-ID: <DABF94C5-A8D6-4779-8097-80C16F7F7E6C@freeswitch.org>

What I think would be neat is to have a perl script to parse the XML  
cdr and spit out a graphic of the call path... now that would be neat.
/b

On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote:

> Thanks Michael,
>
> I'm going to use XML, since I don't really know what variables I want.
> Another problem with CSV is that many people parse them with regular
> expressions and scripts break when you add a new column.




From msc at freeswitch.org  Fri Dec 12 12:22:03 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 12:22:03 -0800
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <1229112873.4100.23.camel@gathern.lan>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
Message-ID: <87f2f3b90812121222v7bd1868id093fa6dbe6c368e@mail.gmail.com>

On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu <alex at sinapticode.ro> wrote:
> Thanks Michael,
>
> I'm going to use XML, since I don't really know what variables I want.
> Another problem with CSV is that many people parse them with regular
> expressions and scripts break when you add a new column.
>

This is true. If you build a proper parser for your XML it will easily
be able to handle new channel variables.
-MC



From gmaruzz at celliax.org  Fri Dec 12 12:25:32 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 12 Dec 2008 21:25:32 +0100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
Message-ID: <7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>

Hi there,

you have to use the "default" ALSA audio device to share it, and to
have it automatically format and rate converted.

the "default" ALSA device is not the default portaudio device (not in
the portaudio version used currently by FS).

You have to find out what device id it has under portaudio.

But in this specific case, no device at all was found.

So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA


Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Fri, Dec 12, 2008 at 8:58 PM, Michael Collins <msc at freeswitch.org> wrote:
> Jason,
>
> If I understand correctly software other than PA can lock up the sound
> card so that PA doesn't "see" it. That might explain why PA reports
> number of devices = 0. Could you check to see if possibly something
> else has control of your sound card, perhaps ALSA? Turn off anything
> that might use the sound card and then restart FS to see if PA can
> then detect your device.
>
> -MC
>
> On Fri, Dec 12, 2008 at 1:38 AM, Jason White <jason at jasonjgw.net> wrote:
>> I am new to FreeSWITCH; hence this is the first of what will probably be a
>> number of questions as I learn.
>>
>> I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
>> 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.
>>
>> Whenever I try to load the portaudio module I get the following in the logs. I
>> haven't changed anything in the default portaudio configuration that comes
>> with FreeSWITCH.
>>
>> PortAudio version number = 1899
>> PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
>> Number of devices = 0
>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
>> input
>>  device!
>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
>> input
>>  device!
>> 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
>> switch_loadable_module_l
>> oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so
>>
>> Other software that uses portaudio is known to work. I would expect FreeSWITCH
>> to detect my Alsa sound devices.
>>
>> Suggestions welcome.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From sicfslist at gmail.com  Fri Dec 12 12:28:02 2008
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Fri, 12 Dec 2008 14:28:02 -0600
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <87f2f3b90812121222v7bd1868id093fa6dbe6c368e@mail.gmail.com>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
	<87f2f3b90812121222v7bd1868id093fa6dbe6c368e@mail.gmail.com>
Message-ID: <35b355e90812121228k5b10aa57k2928849135f5afdb@mail.gmail.com>

Are there any good examples floating around of XML parsers for this to dump
to MySQL?

On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins <msc at freeswitch.org> wrote:

> On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu <alex at sinapticode.ro>
> wrote:
> > Thanks Michael,
> >
> > I'm going to use XML, since I don't really know what variables I want.
> > Another problem with CSV is that many people parse them with regular
> > expressions and scripts break when you add a new column.
> >
>
> This is true. If you build a proper parser for your XML it will easily
> be able to handle new channel variables.
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alex at sinapticode.ro  Fri Dec 12 12:27:25 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Fri, 12 Dec 2008 22:27:25 +0200
Subject: [Freeswitch-users] Configuring FreeSwitch
In-Reply-To: <494129E2.5010602@freeswitch.org>
References: <1228987267.4843.6.camel@gathern.lan>
	<494129E2.5010602@freeswitch.org>
Message-ID: <1229113645.4100.34.camel@gathern.lan>

On Thu, 2008-12-11 at 09:55 -0500, Raymond Chandler wrote:
> 
> i think i answered all of this for you on irc yesterday....
> 

Yes you did, thanks for your help.
I'm a total newbie, but the good news is that I'm almost finished with
my setup. FS is great :)

> use the bridge dialplan app to dial by ip similar to the following:
>     <action application="bridge" 
> data="sofia/${use_profile}/number at ip.address"/>

I'm using "originate" initially. And I think I did something stupid. Is
there anything wrong with the following code ...

var new_session = new Session();
new_session.originate(session, URL);
bridge(session, new_session);

> http://wiki.freeswitch.org/wiki/Sofia#Syntax might also help you out
> a 
> little

It worked great. Thanks.




From gmaruzz at celliax.org  Fri Dec 12 12:30:16 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 12 Dec 2008 21:30:16 +0100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
Message-ID: <7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>

Sorry, the previous one was sent by mistake.

This one is complete:


Hi there,

you have to use the "default" ALSA audio device to share it, and to
have it automatically format and rate converted.

the "default" ALSA device is not the default portaudio device (not in
the portaudio version used currently by FS).

You have to find out what device id it has under portaudio.

But in this specific case, no device at all was found.

So, maybe portaudio was not commpiled with ALSA support (do you have
the ALSA development library installed?).

Also, after recompiling portaudio and mod_portaudio, you can launch FS
giving it the PA_ALSA_PLUGHW=1 environment variable, so portaudio will
use the plughw devices (that are automatically converted to the
desired rate/format) and not the raw devices.

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Fri, Dec 12, 2008 at 9:25 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Hi there,
>
> you have to use the "default" ALSA audio device to share it, and to
> have it automatically format and rate converted.
>
> the "default" ALSA device is not the default portaudio device (not in
> the portaudio version used currently by FS).
>
> You have to find out what device id it has under portaudio.
>
> But in this specific case, no device at all was found.
>
> So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> Company : Celliax
> Website: www.celliax.org
> Address : via Pierlombardo 9, 20135 Milano
> Country/Territory : Italy
> Business Email: gmaruzz at celliax dot org
> Cell : 39-347-2665618
> Fax : 39-02-87390039
>
>
>
>
> On Fri, Dec 12, 2008 at 8:58 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Jason,
>>
>> If I understand correctly software other than PA can lock up the sound
>> card so that PA doesn't "see" it. That might explain why PA reports
>> number of devices = 0. Could you check to see if possibly something
>> else has control of your sound card, perhaps ALSA? Turn off anything
>> that might use the sound card and then restart FS to see if PA can
>> then detect your device.
>>
>> -MC
>>
>> On Fri, Dec 12, 2008 at 1:38 AM, Jason White <jason at jasonjgw.net> wrote:
>>> I am new to FreeSWITCH; hence this is the first of what will probably be a
>>> number of questions as I learn.
>>>
>>> I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
>>> 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.
>>>
>>> Whenever I try to load the portaudio module I get the following in the logs. I
>>> haven't changed anything in the default portaudio configuration that comes
>>> with FreeSWITCH.
>>>
>>> PortAudio version number = 1899
>>> PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
>>> Number of devices = 0
>>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
>>> input
>>>  device!
>>> 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
>>> input
>>>  device!
>>> 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
>>> switch_loadable_module_l
>>> oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so
>>>
>>> Other software that uses portaudio is known to work. I would expect FreeSWITCH
>>> to detect my Alsa sound devices.
>>>
>>> Suggestions welcome.
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From msc at freeswitch.org  Fri Dec 12 12:29:22 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 12:29:22 -0800
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <DABF94C5-A8D6-4779-8097-80C16F7F7E6C@freeswitch.org>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
	<DABF94C5-A8D6-4779-8097-80C16F7F7E6C@freeswitch.org>
Message-ID: <87f2f3b90812121229o6cbf1fb7x58d9fceed3b5238c@mail.gmail.com>

On Fri, Dec 12, 2008 at 12:21 PM, Brian West <brian at freeswitch.org> wrote:
> What I think would be neat is to have a perl script to parse the XML
> cdr and spit out a graphic of the call path... now that would be neat.
> /b

I think that is a great idea. I was kicking that around as an add-on
feature to a simple CDR database. For example, when browsing the db
for calls, you could click a link that says "view call path" and it
would print a nice purty graph/chart of the call flow. I'll put that
on my rainy-day list...

-MC

>
> On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote:
>
>> Thanks Michael,
>>
>> I'm going to use XML, since I don't really know what variables I want.
>> Another problem with CSV is that many people parse them with regular
>> expressions and scripts break when you add a new column.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From msc at freeswitch.org  Fri Dec 12 12:32:44 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 12:32:44 -0800
Subject: [Freeswitch-users] CDR logs - adding a custom field
In-Reply-To: <35b355e90812121228k5b10aa57k2928849135f5afdb@mail.gmail.com>
References: <1229107046.4100.18.camel@gathern.lan>
	<87f2f3b90812121150v4b2f86bp6e7fce3e0baff403@mail.gmail.com>
	<1229112873.4100.23.camel@gathern.lan>
	<87f2f3b90812121222v7bd1868id093fa6dbe6c368e@mail.gmail.com>
	<35b355e90812121228k5b10aa57k2928849135f5afdb@mail.gmail.com>
Message-ID: <87f2f3b90812121232l48ee68d5wf48565bbb5ea28b2@mail.gmail.com>

I don't know about "good" examples. I just hacked together a perl
script to extract the very specific elements for my application. If
anyone out there has a sample XML-to-db parser that would be very
welcomed...
-MC

On Fri, Dec 12, 2008 at 12:28 PM, Shelby Ramsey <sicfslist at gmail.com> wrote:
> Are there any good examples floating around of XML parsers for this to dump
> to MySQL?
>
> On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu <alex at sinapticode.ro>
>> wrote:
>> > Thanks Michael,
>> >
>> > I'm going to use XML, since I don't really know what variables I want.
>> > Another problem with CSV is that many people parse them with regular
>> > expressions and scripts break when you add a new column.
>> >
>>
>> This is true. If you build a proper parser for your XML it will easily
>> be able to handle new channel variables.
>> -MC
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From pedro2263 at gmail.com  Fri Dec 12 13:47:09 2008
From: pedro2263 at gmail.com (Pedro .)
Date: Fri, 12 Dec 2008 15:47:09 -0600
Subject: [Freeswitch-users] Cepstral SDK
Message-ID: <28b3653a0812121347i740fae06w73065605bc7b6eba@mail.gmail.com>

Hi,

I'm trying to integrate Cepstral TTS I read in the wiki that I need
Ceptral's SDK to compile the mod_ceptral, can somebody tell me where can I
get the trial version of this SDK?.

Thanks.
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From brian at freeswitch.org  Fri Dec 12 14:14:37 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 16:14:37 -0600
Subject: [Freeswitch-users] Cepstral SDK
In-Reply-To: <28b3653a0812121347i740fae06w73065605bc7b6eba@mail.gmail.com>
References: <28b3653a0812121347i740fae06w73065605bc7b6eba@mail.gmail.com>
Message-ID: <726FFD6A-7E1D-4BE9-A96C-2D885BDF5931@freeswitch.org>

If you're on linux you need to go download and install any voice.  If  
you're on windows I have to forward your request to Cepstral to get  
the SDK for windows.

/b

On Dec 12, 2008, at 3:47 PM, Pedro . wrote:

> Hi,
>
> I'm trying to integrate Cepstral TTS I read in the wiki that I need  
> Ceptral's SDK to compile the mod_ceptral, can somebody tell me where  
> can I get the trial version of this SDK?.
>
> Thanks.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jason at jasonjgw.net  Fri Dec 12 15:24:03 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 10:24:03 +1100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
Message-ID: <20081212232403.GA7667@jdc.jasonjgw.net>

On Fri, Dec 12, 2008 at 09:30:16PM +0100, Giovanni Maruzzelli wrote:
 
> But in this specific case, no device at all was found.
> 
> So, maybe portaudio was not commpiled with ALSA support (do you have
> the ALSA development library installed?).

Yes, and in any case the version of PortAudio which is installed came from the
Debian package.

Does FreeSWITCH support PortAudio 19? If not, maybe there are API differences.
> 
> Also, after recompiling portaudio and mod_portaudio, you can launch FS
> giving it the PA_ALSA_PLUGHW=1 environment variable, so portaudio will
> use the plughw devices (that are automatically converted to the
> desired rate/format) and not the raw devices.

I'll try that.

To answer another question that arose in this thread, I have no other software
currently using the audio devices. Alsa is known to work, as is other software
that accesses the Alsa devices with PortAudio.




From frank at impactfax.com  Fri Dec 12 15:51:35 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 12 Dec 2008 18:51:35 -0500
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
Message-ID: <026101c95cb4$91ad1d40$33014c0a@ws4>

Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call?
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From brian at freeswitch.org  Fri Dec 12 15:54:44 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2008 17:54:44 -0600
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <026101c95cb4$91ad1d40$33014c0a@ws4>
References: <026101c95cb4$91ad1d40$33014c0a@ws4>
Message-ID: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>

sched_api  (hint uuid_send_dtmf)

API CALL [sched_api()] output:
-ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>


/b

On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:

> Is there a way to schedule a certain DTMF tone to be played into a  
> bridge (both a and b legs) after a scheduled number of seconds into  
> the call?
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Fri Dec 12 16:50:04 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 11:50:04 +1100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081212232403.GA7667@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
Message-ID: <20081213005004.GA8393@jdc.jasonjgw.net>

This is interesting...

I wrote the following test.


#include<stdio.h>
#include <portaudio.h>
int main(int argc, char **argv) 
{
  Pa_Initialize();
  printf("Number of devices: %d\n",Pa_GetDeviceCount());
  Pa_Terminate();
}

then I compiled and executed it:
gcc -o pa_test pa_test.c -lportaudio
./portaudio

Number of devices: 10

>From what I can see, the code in mod_portaudio.c is using exactly the same API
call, but it seems to be returning 0 in that case.




From jason at jasonjgw.net  Fri Dec 12 17:35:38 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 12:35:38 +1100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081213005004.GA8393@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
Message-ID: <20081213013538.GA16338@jdc.jasonjgw.net>

With apologies for the noise on the list, I just realized that FreeSWITCH is
building its own version of PortAudio.

I can confirm that Alsa is being detected and support for it included.

So, there must be some difference between the version of PortAudio that comes
with FreeSWITCH, and the version installed in my /usr/lib, such that the
FreeSWITCH version fails to detect my Alsa devices.




From ack at telefonica.net  Fri Dec 12 17:58:01 2008
From: ack at telefonica.net (Angel Carpintero)
Date: Sat, 13 Dec 2008 02:58:01 +0100
Subject: [Freeswitch-users] missing 3 seconds of audio on bridge calls
In-Reply-To: <1228875010.2477.67.camel@develop4>
References: <1228352588.25709.42.camel@develop4>
	<191c3a030812032012g47ec04a9j965988d8b67e7854@mail.gmail.com>
	<1228365981.25709.60.camel@develop4>
	<191c3a030812040734s4f514f42s9a30a48c93709fd5@mail.gmail.com>
	<1228875010.2477.67.camel@develop4>
Message-ID: <1229133482.10071.59.camel@develop4>

Thanks again Anthony ! 

You fixed the issue with DTMF i had reported :

http://jira.freeswitch.org/browse/FSCORE-251



Chris Danielson added to Wiki a nice page collecting these issues with
Sonus :

http://wiki.freeswitch.org/wiki/RTP_Issues


Cheers,

El mi?, 10-12-2008 a las 03:10 +0100, Angel Carpintero escribi?:
> Thanks Anthony , you did a great work ! this is fixed in svn r10691.
> 
> Some notes for people using Sonus and L3 as was my case :
> 
> in var.xml in some scenario you may need :
> 
> <X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/>
> 
> in sip_profiles/internal.xml :
> 
> <param name="rtp-rewrite-timestamps" value="true"/>
> 
> might help for some people with rtp issues :
> 
> <param name="rtp-timer-name" value="none"/>
> 
> If you have issues with DTMF and timestamps add also :
> 
> <param name="pass-rfc2833" value="true"/>
> 
> I've a little issues with DTMF from VOIP , i i'll figure out can could
> be the issue , from PSTN all works like a charm :)
> 
> Cheers,
> 
> El jue, 04-12-2008 a las 09:34 -0600, Anthony Minessale escribi?:
> > most likely it's because during the time you are dong artificial
> > ringback the other side is not doing RTP right.
> > 
> > When the call is answered we flush the rtp buffer and your missing
> > audio is probably flushed with it.
> > so you can choose to have a 3 second delay or erase the 3 seconds as
> > it does now.
> > 
> > This is a known problem with sonus which has been proven to build up
> > an audio delay during the time
> > you are waiting for the call to answer.  I'm sure you prefer the way
> > it is to a large audio delay.
> > 
> > 
> > 
> > On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <ack at telefonica.net>
> > wrote:
> >         No TDM , all is SIP :
> >         
> >         
> >         PSTN ---> Sip Proxy_A --> FS ( brigde )
> >         ringback/transfer_ringback
> >         -> Sip Proxy_B --> PSTN
> >         
> >         
> >         In logfile i think you can get some details about Media
> >         Gateways
> >         ( Sonus ) PSTN inbound / outbound is provided by Level3.
> >         
> >         I can get a capture of a call if you want, in capture the
> >         audio is not
> >         missing, issue with :
> >         
> >         - rtp buffer ?
> >         - Sonus ?
> >         
> >         Let me know anything you need so i can provide a log or create
> >         a new
> >         scenario.
> >         
> >         
> >         Thanks,
> >         
> >         El mi?, 03-12-2008 a las 22:12 -0600, Anthony Minessale
> >         escribi?:
> >         
> >         > what does PSTN represent?
> >         >
> >         > I know what the PSTN is but how are you reaching it?
> >         > is it TDM, SIP etc... what gateway type other details.
> >         >
> >         >
> >         > On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero
> >         <ack at telefonica.net>
> >         > wrote:
> >         >         Hi guys,
> >         >
> >         >          I've a strange issue with FS , version svn
> >         -r10584 ,
> >         >         when FS bridges a call first 3 seconds of audio are
> >         missing ,
> >         >         looks that
> >         >         only happens on PSTN calls and using ringback or
> >         >         transfer_ringback. This
> >         >         only happens in calls from PSTN , not from VOIP.
> >         Some
> >         >         scenarios i tried
> >         >         to isolate this issue :
> >         >
> >         >
> >         >         - Issue
> >         >
> >         >         PSTN --> FS ( brigde ) ringback/transfer_ringback ->
> >         PSTN
> >         >
> >         >         - Good setting bypass_media before run bridge but i
> >         need rtp
> >         >         in FS path
> >         >
> >         >         PSTN --> FS ( brigde ) ringback/transfer_ringback ->
> >         PSTN
> >         >
> >         >         - Good
> >         >
> >         >         PSTN --> FS ( brigde ) WITHOUT
> >         ringback/transfer_ringback ->
> >         >         PSTN
> >         >
> >         >         - Always good
> >         >
> >         >         VOIP --> FS ( brigde ) -> PSTN
> >         >
> >         >
> >         >         Dialplan has nothing wrong ( i guess ):
> >         >
> >         >         <extension name="Transfers">
> >         >            <condition field="destination_number"
> >         >         expression="^1??XXXXXXXXXX$">
> >         >              <action application="answer"/>
> >         >              <action application="speak" data="cepstral|
> >         Allison-8kHz|
> >         >         blah"/>
> >         >              <action application="set"
> >         >         data="hangup_after_bridge=false"/>
> >         >              <action application="set"
> >         data="playback_terminators=#"/>
> >         >              <action application="set" data="ringback=
> >         $${us-ring}"/>
> >         >              <action application="set"
> >         data="transfer_ringback=
> >         >         $${hold_music}"/>
> >         >              <action application="set"
> >         data="effective_caller_id_name=
> >         >         ${caller_id_name}"/>
> >         >              <action application="set"
> >         >         data="effective_caller_id_number=
> >         >         ${caller_id_number}"/>
> >         >              <action application="set"
> >         data="originate_timeout=30"/>
> >         >              <action application="set"
> >         data="call_timeout=30"/>
> >         >              <action application="bridge"
> >         >         data="sofia/default/18008226235 at PSTN_GW"/>
> >         >              <action application="speak" data="cepstral|
> >         Allison-8kHz|
> >         >         Transfer
> >         >         finished"/>
> >         >              <action application="hangup"/>
> >         >            </condition>
> >         >          </extension>
> >         >
> >         >
> >         >
> >         >         Any ideas ?
> >         >
> >         >         Attached log of FS ( F8 from console ).
> >         >
> >         >
> >         >         Thanks in advance !
> >         >
> >         >         --
> >         >         Angel Carpintero
> >         >         ack ( at ) telefonica ( dot ) net
> >         >
> >         >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF
> >         AC2C CA61
> >         >         6EF1 B90D
> >         >
> >         >
> >         >
> >         
> >         > --
> >         > Anthony Minessale II
> >         >
> >         > FreeSWITCH http://www.freeswitch.org/
> >         > ClueCon http://www.cluecon.com/
> >         >
> >         > AIM: anthm
> >         > MSN:anthony_minessale at hotmail.com
> >         > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >         > IRC: irc.freenode.net #freeswitch
> >         >
> >         > FreeSWITCH Developer Conference
> >         > sip:888 at conference.freeswitch.org
> >         > iax:guest at conference.freeswitch.org/888
> >         > googletalk:conf+888 at conference.freeswitch.org
> >         > pstn:213-799-1400
> >         
> >         
> >         --
> >         
> >         Angel Carpintero
> >         ack ( at ) telefonica ( dot ) net
> >         
> >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
> >         6EF1 B90D
> >         
> >         
> >         
> >         _______________________________________________
> >         Freeswitch-users mailing list
> >         Freeswitch-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >         
> > 
> > 
> > 
> > -- 
> > Anthony Minessale II
> > 
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > 
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> > 
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:213-799-1400
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

"No basta saber, hay que aplicar lo que se sabe; 
no basta querer hacerlas cosas, hay que hacerlas".

"Knowing is not enough; we must apply. 
 Willing is not enough; we must do"

 Johann Wolfgang von Goethe
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From jason at jasonjgw.net  Fri Dec 12 18:01:36 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 13:01:36 +1100
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081213013538.GA16338@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
	<20081213013538.GA16338@jdc.jasonjgw.net>
Message-ID: <20081213020136.GA16597@jdc.jasonjgw.net>

The problem is now solved.

It turned out to be permissions: the freeswitch user wasn't added to the audio
group in /etc/group, hence didn't have permission to interrogate the audio
devices.

Perhaps a future version of the Debian package could address this, or at least
it should be noted somewhere.




From msc at freeswitch.org  Fri Dec 12 18:25:58 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 18:25:58 -0800
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081213020136.GA16597@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
	<20081213013538.GA16338@jdc.jasonjgw.net>
	<20081213020136.GA16597@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812121825t531a60b6k60a3e57796338578@mail.gmail.com>

Jason,

Thanks for troubleshooting this! At the very least I will add a note
to the PA section of the wiki.
-MC

On Fri, Dec 12, 2008 at 6:01 PM, Jason White <jason at jasonjgw.net> wrote:
> The problem is now solved.
>
> It turned out to be permissions: the freeswitch user wasn't added to the audio
> group in /etc/group, hence didn't have permission to interrogate the audio
> devices.
>
> Perhaps a future version of the Debian package could address this, or at least
> it should be noted somewhere.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From frank at impactfax.com  Fri Dec 12 19:37:20 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 12 Dec 2008 22:37:20 -0500
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>
Message-ID: <02b601c95cd4$1b093460$33014c0a@ws4>

Not much written in the wiki on this.  Also searched the list and not
much on either sched_api or uuid_send_dtmf.
So from an xml dialplan, can sched_api as an application?
Is there any way to have the time offset reference the point at which
the call started ?  ie. When the called party answers?
 
Ultimately, I was trying to insert some xml into my dial plan that would
play a dtmf tone 10 seconds after the called party picked up the phone.
But from the little that has been written so far that I can find, it is
not clear to me how to piece this together.  Am I being dense and
missing anything that has already been written?
 
/f
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West


sched_api  (hint uuid_send_dtmf)
 
API CALL [sched_api()] output:
-ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>
 
 
/b
 
On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:



Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call?
_______________________________________________
 
 
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From msc at freeswitch.org  Fri Dec 12 20:21:27 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 20:21:27 -0800
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <02b601c95cd4$1b093460$33014c0a@ws4>
References: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>
	<02b601c95cd4$1b093460$33014c0a@ws4>
Message-ID: <87f2f3b90812122021y2979d21anadab85a70ed4717e@mail.gmail.com>

Frank,

I'm sure this is possible. Please give me a little bit to look into
this. I'm going to see if I can lab it up and give you a sample
dialplan. Also, thanks for the heads up on the wiki not having this
information. I will put that on my not-so-short wiki todo list.

Thanks,
MC

On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact <frank at impactfax.com> wrote:
> Not much written in the wiki on this.  Also searched the list and not much
> on either sched_api or uuid_send_dtmf.
>
> So from an xml dialplan, can sched_api as an application?
>
> Is there any way to have the time offset reference the point at which the
> call started ?  ie. When the called party answers?
>
>
>
> Ultimately, I was trying to insert some xml into my dial plan that would
> play a dtmf tone 10 seconds after the called party picked up the phone.  But
> from the little that has been written so far that I can find, it is not
> clear to me how to piece this together.  Am I being dense and missing
> anything that has already been written?
>
>
>
> /f
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
>
> sched_api  (hint uuid_send_dtmf)
>
>
>
> API CALL [sched_api()] output:
>
> -ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>
>
>
>
>
>
> /b
>
>
>
> On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:
>
> Is there a way to schedule a certain DTMF tone to be played into a bridge
> (both a and b legs) after a scheduled number of seconds into the call?
>
> _______________________________________________
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Dec 12 22:39:21 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 22:39:21 -0800
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <02b601c95cd4$1b093460$33014c0a@ws4>
References: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>
	<02b601c95cd4$1b093460$33014c0a@ws4>
Message-ID: <87f2f3b90812122239y127f35d6r80cd1286b7dddae8@mail.gmail.com>

Frank,

I found a simple way to handle this scenario. I decided just to create
a small Lua script that would do the job. It's committed in latest
trunk. Look in src/scripts/contrib/mcollins for uuid_send_dtmf.lua. It
has comments on how to call it, including a sample dp call.

The way I would use this in your scenario is to setup a destination
using the execute_on_answer variable.
http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer

Have the destination be an extension that does something like this:
<extension...>
  <condition...>
    <action application="answer"/>
    <action application="lua" data="uuid_send_dtmf ${uuid} 10 123"/>
    ...rest of diaplan...
  </condition>
</extension>

The 10 means ten seconds, the 123 means send the dtmf digits 1,2,3 in
order. You can tinker with the settings as you see fit.

Please let me know how it goes. BTW, be sure to put the Lua script in
/usr/local/freeswitch/scripts or specify the complete path name when
calling the lua app in the dialplan.

-MC

On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact <frank at impactfax.com> wrote:
> Not much written in the wiki on this.  Also searched the list and not much
> on either sched_api or uuid_send_dtmf.
>
> So from an xml dialplan, can sched_api as an application?
>
> Is there any way to have the time offset reference the point at which the
> call started ?  ie. When the called party answers?
>
>
>
> Ultimately, I was trying to insert some xml into my dial plan that would
> play a dtmf tone 10 seconds after the called party picked up the phone.  But
> from the little that has been written so far that I can find, it is not
> clear to me how to piece this together.  Am I being dense and missing
> anything that has already been written?
>
>
>
> /f
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
>
> sched_api  (hint uuid_send_dtmf)
>
>
>
> API CALL [sched_api()] output:
>
> -ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>
>
>
>
>
>
> /b
>
>
>
> On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:
>
> Is there a way to schedule a certain DTMF tone to be played into a bridge
> (both a and b legs) after a scheduled number of seconds into the call?
>
> _______________________________________________
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Dec 12 22:41:00 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Dec 2008 22:41:00 -0800
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <87f2f3b90812122021y2979d21anadab85a70ed4717e@mail.gmail.com>
References: <7A216626-D8A0-4CE1-9C82-99E6CCB9480D@freeswitch.org>
	<02b601c95cd4$1b093460$33014c0a@ws4>
	<87f2f3b90812122021y2979d21anadab85a70ed4717e@mail.gmail.com>
Message-ID: <87f2f3b90812122241t5dfa8bdeif15299684666c12e@mail.gmail.com>

On Fri, Dec 12, 2008 at 8:21 PM, Michael Collins <msc at freeswitch.org> wrote:
> Frank,
>
> I'm sure this is possible. Please give me a little bit to look into
> this. I'm going to see if I can lab it up and give you a sample
> dialplan. Also, thanks for the heads up on the wiki not having this
> information. I will put that on my not-so-short wiki todo list.
>
> Thanks,
> MC
>
> On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact <frank at impactfax.com> wrote:
>> Not much written in the wiki on this.  Also searched the list and not much
>> on either sched_api or uuid_send_dtmf.
>>
>> So from an xml dialplan, can sched_api as an application?
>>
>> Is there any way to have the time offset reference the point at which the
>> call started ?  ie. When the called party answers?
>>
>>
>>
>> Ultimately, I was trying to insert some xml into my dial plan that would
>> play a dtmf tone 10 seconds after the called party picked up the phone.  But
>> from the little that has been written so far that I can find, it is not
>> clear to me how to piece this together.  Am I being dense and missing
>> anything that has already been written?
>>
>>
>>
>> /f
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
>> West
>>
>> sched_api  (hint uuid_send_dtmf)
>>
>>
>>
>> API CALL [sched_api()] output:
>>
>> -ERR Invalid syntax. USAGE: [+@]<time> <group_name> <command_string>
>>
>>
>>
>>
>>
>> /b
>>
>>
>>
>> On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:
>>
>> Is there a way to schedule a certain DTMF tone to be played into a bridge
>> (both a and b legs) after a scheduled number of seconds into the call?
>>
>> _______________________________________________
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From mszlazak at aol.com  Sat Dec 13 00:05:55 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Sat, 13 Dec 2008 03:05:55 -0500
Subject: [Freeswitch-users] Confirmation checking on VAD is needed.
Message-ID: <8CB2B10C221A1AB-4C4-1AB5@WEBMAIL-MY18.sysops.aol.com>

I need confirmation on whether VAD is disabled in the FS default configuration. I've looked at conf\sip_profiles\internal.xml and the relevant section was the following which seems to indicate that VAD is off by default since I don't see VAD value="none".

??? <!-- VAD choose one (out is a good choice); -->
??? <!-- <param name="vad" value="in"/> -->
??? <!-- <param name="vad" value="out"/> -->
??? <!-- <param name="vad" value="both"/> -->
??? <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
??? <!--all inbound reg will look in this domain for the users -->
??? <!--<param name="force-register-domain" value="cluecon.com"/>-->
??? <!-- disable register and transfer which may be undesirable in a public switch -->
??? <!--<param name="disable-transfer" value="true"/>-->
??? <!--<param name="disable-register" value="true"/>-->

However, is there a way to confirm this in the FS console?
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From jason at jasonjgw.net  Sat Dec 13 00:46:43 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Dec 2008 19:46:43 +1100
Subject: [Freeswitch-users] Confirmation checking on VAD is needed.
In-Reply-To: <8CB2B10C221A1AB-4C4-1AB5@WEBMAIL-MY18.sysops.aol.com>
References: <8CB2B10C221A1AB-4C4-1AB5@WEBMAIL-MY18.sysops.aol.com>
Message-ID: <20081213084643.GA2378@jdc.jasonjgw.net>

On Sat, Dec 13, 2008 at 03:05:55AM -0500, mszlazak at aol.com wrote:
> I need confirmation on whether VAD is disabled in the FS default configuration. I've looked at conf\sip_profiles\internal.xml and the relevant section was the following which seems to indicate that VAD is off by default since I don't see VAD value="none".
> 

This is all commented out, and from a quick look at the code in the sofia SIP
module, VAD won't be enabled unless you explicitly set it to "in" or "out".




From faisalmaqsoodi at yahoo.com  Sat Dec 13 01:05:22 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sat, 13 Dec 2008 01:05:22 -0800 (PST)
Subject: [Freeswitch-users] Maintaining call detail record
Message-ID: <831510.71942.qm@web30702.mail.mud.yahoo.com>

How can i interface fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked on xml cdr but it contains too much info, more than i need and in a format which is not easily understandable. I also tried perl coding mentioned on the link at the bottom of the page
http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r displayed during its execution. Is there any easy method for that. Plz help me. 
?????????????????????????????????????????????????????????????????????????????????????? faisal



      
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From faisalmaqsoodi at yahoo.com  Sat Dec 13 01:44:38 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sat, 13 Dec 2008 01:44:38 -0800 (PST)
Subject: [Freeswitch-users] Maintaining call detail record
Message-ID: <614188.90296.qm@web30706.mail.mud.yahoo.com>

                        
                        

if (typeof YAHOO == "undefined") {
 var YAHOO = {};
}
YAHOO.Shortcuts = YAHOO.Shortcuts || {};
YAHOO.Shortcuts.hasSensitiveText = false;
YAHOO.Shortcuts.sensitivityType = [];
YAHOO.Shortcuts.doUlt = false;
YAHOO.Shortcuts.location = "us";
YAHOO.Shortcuts.document_id = 0;
YAHOO.Shortcuts.document_type = "";
YAHOO.Shortcuts.document_title = "Maintaining call detail record";
YAHOO.Shortcuts.document_publish_date = "";
YAHOO.Shortcuts.document_author = "faisalmaqsoodi at yahoo.com";
YAHOO.Shortcuts.document_url = "";
YAHOO.Shortcuts.document_tags = "";
YAHOO.Shortcuts.document_language = "english";
YAHOO.Shortcuts.annotationSet = {
"lw_1229161419_0": {
"text": "caller id",
"extended": 0,
"startchar": 187,
"endchar": 195,
"start": 187,
"end": 195,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 0.2054,
"relScore": 5.33095,
"type": ["shortcuts:/concept"],
"category": ["CONCEPT"],
"wikiId": "Caller_ID",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked",
"metaData": {
"visible": "true"
}
 },
"lw_1229161419_1": {
"text": "http://wiki.freeswitch.org/wiki/Mod_cdr_csv,",
"extended": 0,
"startchar": 425,
"endchar": 468,
"start": 425,
"end": 468,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 1,
"relScore": 0,
"type": ["shortcuts:/us/instance/identifier/URL"],
"category": ["IDENTIFIER"],
"wikiId": "",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "",
"metaData": {
"visible": "true"
}
 }
};
YAHOO.Shortcuts.headerID = "c0b92bae50300eda0a2889f03525f330";

How can i interface fs with mysql in order to maintain calls record like caller id
and time n date of call etc. I ve worked on xml cdr but it contains too
much info, more than i need and in a format which is not easily
understandable. I also tried perl coding mentioned on the link at the
bottom of the page
http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r displayed during its execution. Is there any easy method for that. Plz help me.
 
?????????????????????????????????????????????????????????????????????????????????????? faisal


      
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From dalechase at yahoo.com  Sat Dec 13 11:03:34 2008
From: dalechase at yahoo.com (dalechase at yahoo.com)
Date: Sat, 13 Dec 2008 11:03:34 -0800 (PST)
Subject: [Freeswitch-users] config help: openzap and T1 A102u
References: <603978.74977.qm@web36108.mail.mud.yahoo.com>
	<004001c95c38$04da0210$6d02a8c0@opos20>
Message-ID: <255403.64286.qm@web36104.mail.mud.yahoo.com>

Thanks, that got things going...



----- Original Message ----
From: Evgeniy Zolotov <zolotov at altron.ua>
To: freeswitch-users at lists.freeswitch.org
Sent: Friday, December 12, 2008 1:00:01 AM
Subject: Re: [Freeswitch-users] config help: openzap and T1 A102u

  Did you try ./wanrouter start before starting FreeSWITCH ?


      



From chavpaskov at shaw.ca  Sat Dec 13 12:24:25 2008
From: chavpaskov at shaw.ca (Chav Paskov)
Date: Sat, 13 Dec 2008 12:24:25 -0800
Subject: [Freeswitch-users] how to handle  returned sip 302 dialplan
Message-ID: <494419F9.6090304@shaw.ca>

Good morning everybody,

I would like to know what would be the best  way to process the info in 
returned back sip 302  message.
Let me be a bit more specific.
i'm sending INVITE as pasted below to  z.z.z.z

*INVITE* sip:17402740539 at z.z.z.z:5000 SIP/2.0^M
Via: SIP/2.0/UDP z.z.z.z:5000;branch=z9hG4bK215b.e6ae7e74.0^M
Via: SIP/2.0/UDP 
y.y.y.y:5060;branch=z9hG4bKaa6257694041475e0659eeb0019a0bbc^M
*From: "Bob Caller" <sip:6144102508 at y.y.y.y:5060>;tag=CCC7.1F37^M
To: <sip:17402740539 at x.x.x.x:5060>^M
*Call-ID: 15821337-3367080415-112396 at msc-01.voice.example.net^M
CSeq: 1 INVITE^M
Contact: <sip:6144102508 at y.y.y.y:5060>^M
Content-Type: application/sdp^M
Max-forwards: 69^M
Session-expires: 3600;Refresher=uac^M
Supported: timer^M
Content-Length:   200^M
^M
1192 v=0^M
o=iserver 16123 16124 IN IP4 y.y.y.y^M
s=sip call^M
c=IN IP4 64.112.188.81^M
t=0 0^M
m=audio 55824 RTP/AVP 18 0 101^M
a=sendrecv^M
a=fmtp:101 0-15^M
a=rtpmap:101 telephone-event/8000^M
a=ptime:20^M

the reply from  z.z.z.z back is  sip 302

SIP/2.0 *302 Moved Temporarily*^M
Via: SIP/2.0/UDP z.z.z.z:5000;branch=z9hG4bK215b.e6ae7e74.0^M
Via: SIP/2.0/UDP 
y.y.y.y:5060;branch=z9hG4bKaa6257694041475e0659eeb0019a0bbc^M
From: "Bob Caller" <sip:6144102508 at y.y.y.y:5060>;tag=CCC7.1F37^M
To: <sip:17402740539 at x.x.x.x:5060>;tag=1087415105^M
Call-ID: 15821337-3367080415-112396 at msc-01.voice.example.net^M
CSeq: 1 INVITE^M
*Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
*User-Agent: eXosip/3.1.0^M
Content-Length:  


my question is:
 
Is it possible to send the call to z.z.z.z , receive the SIP 302 , 
process the data in Contact field  and redirect to the new destination 
contained in *Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
*without closing the session.
i red something about   <action application="set" 
data="continue_on_fail=true"/>  but i'm not sure  how to use it.

Any ideas on this matter  will be highly appreciated.
Best Regards
Chav







From msc at freeswitch.org  Sat Dec 13 12:51:12 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 12:51:12 -0800
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <49427CC6.2090407@ewetel.de>
References: <49427CC6.2090407@ewetel.de>
Message-ID: <87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>

On Fri, Dec 12, 2008 at 7:01 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
> AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
> server in a non-root environment.
>
> We experienced a timer problem which led to this FS console error message:
>
> [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries
>
>
> During anylizing this we found that q921 T203 is never reset when link
> is in state "Multiple Frame Mode Established" and SABME frames are
> received by FS. So it must timeout regardless if SABME frames are
> received or not.
> Additionally we found that the default T203 value (10 sec) was too short
> for AVAYA (it has to be >=19 sec)
>
> To fix the problem we changed two things in q921.c:
>
> Change T203 default value from 10 sec to 20000 sec
> Line 406: trunk->T203Timeout = 20000;
>
> Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
> received SABME frame
> Line 1996: Q921T203TimerReset(trunk, tei);
>
> After recompiling FS the Error disapeared. Next week we will do some
> calls over the link to make sure there are no other side effects.
>
> Is it planned to make the q921 timeouts configurable in openzap.conf or
> in openzap.conf.xml?
>
> best regards
> Helmut
>
>
> PS: My openzap configs:
>
> openzap.conf
>
> [span wanpipe PRI_1]
> trunk_type => E1
> b-channel => 1:1-15
> d-channel => 1:16
> b-channel => 1:17-31
>
>
>
>
> openzap.conf.xml
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>  <settings>
>    <param name="debug" value="7"/>
>    <!--<param name="hold-music" value="$${moh_uri}"/>-->
>    <!--<param name="enable-analog-option" value="call-swap"/>-->
>    <!--<param name="enable-analog-option" value="3-way"/>-->
>  </settings>
>   <pri_spans>
>     <span name="PRI_1">
>       <!-- Log Levels: none, alert, crit, err, warning, notice, info,
> debug -->
>       <param name="q921loglevel" value="debug"/>
>       <param name="q931loglevel" value="debug"/>
>       <param name="mode" value="user"/>
>       <param name="dialect" value="Oh this is not my dialect"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="default"/>
>     </span>
>   </pri_spans>
> </configuration>
>
> Very interesting here is, that the "dialect" parameter doesn't seem to
> have an effect on FS. I use that one above without any errors or warning
> and I guess that was not intended.

At this point in the OZ development we've got it set to default to
"national" if the dialect isn't otherwise properly specified. It does
make sense to throw an error if the dialect is not properly specified,
even if we still default to national.

-MC

>
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
> BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
> =geXp
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From astmac at stillnewt.org  Sat Dec 13 13:00:19 2008
From: astmac at stillnewt.org (martin joseph)
Date: Sat, 13 Dec 2008 13:00:19 -0800
Subject: [Freeswitch-users] (no subject)
Message-ID: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>

I am wondering if anyone running OSX 10.4.11 is able to successfully  
build FS Trunk?

If so,  what versions does build.sh report? Please?

I get the following and have issues making it...

Updated to revision 10753.
mail:/usr/src/freeswitch/trunk root# ./bootstrap.sh
bootstrap: checking installation...
bootstrap: autoconf version 2.61 (ok)
bootstrap: automake version 1.10 (ok)
bootstrap: libtool version 1.5.24 (ok)

Thanks for the info,
Marty




From frank at impactfax.com  Sat Dec 13 13:20:16 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 13 Dec 2008 16:20:16 -0500
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <87f2f3b90812122239y127f35d6r80cd1286b7dddae8@mail.gmail.com>
Message-ID: <03c801c95d68$9880ce50$33014c0a@ws4>

Michael,

Got it working.  Just a little simpler then you outlined.
I just added to my xml dialplan this line.

<action application="set" data="execute_on_answer=lua uuid_send_dtmf.lua
${uuid} 20 123"/>

I added this just before the bridge application.

I did this instead of adding an extra extension to transfer to on
answer.  Everything worked well.  The DTMF was played to the calling
party.  Out of curiosity, if we wanted also to play the DTMF to the
called party also, what would we have to give uuid_send_dtmf?
Particularly since we call it before the bridge.

Can uuid_send_dtmf accept anything like "w" for wait or anything else
special for DTMF stuff?

Also, I got an error output to the console when the sched_api was run.
See below.
******
2008-12-13 16:07:28 [NOTICE] switch_cpp.cpp:1050 console_log() apicmd:
sched_api
2008-12-13 16:07:28 [NOTICE] switch_cpp.cpp:1050 console_log() apiarg:
+20 none uuid_send_dtmf 37618e54-c959-11dd-bc73-0923daa880b2 123

2008-12-13 16:07:28 [ERR] switch_cpp.cpp:1050 console_log() Result is
+OK Added: 49751
******
is this ERR anything to worry about even though we got a result ok?

Thanks again for the help.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
 
Frank,

I found a simple way to handle this scenario. I decided just to create
a small Lua script that would do the job. It's committed in latest
trunk. Look in src/scripts/contrib/mcollins for uuid_send_dtmf.lua. It
has comments on how to call it, including a sample dp call.

The way I would use this in your scenario is to setup a destination
using the execute_on_answer variable.
http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer

Have the destination be an extension that does something like this:
<extension...>
  <condition...>
    <action application="answer"/>
    <action application="lua" data="uuid_send_dtmf ${uuid} 10 123"/>
    ...rest of diaplan...
  </condition>
</extension>

The 10 means ten seconds, the 123 means send the dtmf digits 1,2,3 in
order. You can tinker with the settings as you see fit.

Please let me know how it goes. BTW, be sure to put the Lua script in
/usr/local/freeswitch/scripts or specify the complete path name when
calling the lua app in the dialplan.

-MC






From brian at freeswitch.org  Sat Dec 13 13:28:58 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Dec 2008 15:28:58 -0600
Subject: [Freeswitch-users] how to handle  returned sip 302 dialplan
In-Reply-To: <494419F9.6090304@shaw.ca>
References: <494419F9.6090304@shaw.ca>
Message-ID: <9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>

Chav,
	Once the 302 is received by FreeSWITCH it will follow it to the  
contact listed in the 302.  What else are you needing to do?

/b

On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:

> *User-Agent: eXosip/3.1.0^M
> Content-Length:
>
>
> my question is:
>
> Is it possible to send the call to z.z.z.z , receive the SIP 302 ,
> process the data in Contact field  and redirect to the new destination
> contained in *Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
> *without closing the session.
> i red something about   <action application="set"
> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
>
> Any ideas on this matter  will be highly appreciated.
> Best Regards
> Chav
>

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From brian at freeswitch.org  Sat Dec 13 13:30:40 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Dec 2008 15:30:40 -0600
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
Message-ID: <830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>

I don't get what you're saying.. this looks 100% OK.  Also make sure  
you have the dev tools from apple and not automake, autoconf and  
libtool installed via fink,darwinports or any other such method.  If  
you can provide a bit more information maybe I can see what is wrong.

/b

On Dec 13, 2008, at 3:00 PM, martin joseph wrote:

> I get the following and have issues making it...
>
> Updated to revision 10753.
> mail:/usr/src/freeswitch/trunk root# ./bootstrap.sh
> bootstrap: checking installation...
> bootstrap: autoconf version 2.61 (ok)
> bootstrap: automake version 1.10 (ok)
> bootstrap: libtool version 1.5.24 (ok)
>
> Thanks for the info,
> Marty




From mike at jerris.com  Sat Dec 13 13:51:17 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 13 Dec 2008 16:51:17 -0500
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <20081213020136.GA16597@jdc.jasonjgw.net>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
	<20081213013538.GA16338@jdc.jasonjgw.net>
	<20081213020136.GA16597@jdc.jasonjgw.net>
Message-ID: <661B71A1-D73F-43F0-97BA-11D6798D1486@jerris.com>

Please file a bug on Jira.freeswitch.org on this issue so we make sure  
it gets adressef in the debs.

Mike

On Dec 12, 2008, at 9:01 PM, Jason White <jason at jasonjgw.net> wrote:

> The problem is now solved.
>
> It turned out to be permissions: the freeswitch user wasn't added to  
> the audio
> group in /etc/group, hence didn't have permission to interrogate the  
> audio
> devices.
>
> Perhaps a future version of the Debian package could address this,  
> or at least
> it should be noted somewhere.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Sat Dec 13 13:54:16 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 13 Dec 2008 16:54:16 -0500
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>
References: <49427CC6.2090407@ewetel.de>
	<87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>
Message-ID: <C0C90F5C-CC42-451A-AC91-D35BF4846EAE@jerris.com>

Please file a bug on this issue

Mike

On Dec 13, 2008, at 3:51 PM, "Michael Collins" <msc at freeswitch.org>  
wrote:

> On Fri, Dec 12, 2008 at 7:01 AM, Helmut Kuper  
> <helmut.kuper at ewetel.de> wrote:
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hello,
>>
>> I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
>> AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu  
>> 8.04
>> server in a non-root environment.
>>
>> We experienced a timer problem which led to this FS console error  
>> message:
>>
>> [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries
>>
>>
>> During anylizing this we found that q921 T203 is never reset when  
>> link
>> is in state "Multiple Frame Mode Established" and SABME frames are
>> received by FS. So it must timeout regardless if SABME frames are
>> received or not.
>> Additionally we found that the default T203 value (10 sec) was too  
>> short
>> for AVAYA (it has to be >=19 sec)
>>
>> To fix the problem we changed two things in q921.c:
>>
>> Change T203 default value from 10 sec to 20000 sec
>> Line 406: trunk->T203Timeout = 20000;
>>
>> Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
>> received SABME frame
>> Line 1996: Q921T203TimerReset(trunk, tei);
>>
>> After recompiling FS the Error disapeared. Next week we will do some
>> calls over the link to make sure there are no other side effects.
>>
>> Is it planned to make the q921 timeouts configurable in  
>> openzap.conf or
>> in openzap.conf.xml?
>>
>> best regards
>> Helmut
>>
>>
>> PS: My openzap configs:
>>
>> openzap.conf
>>
>> [span wanpipe PRI_1]
>> trunk_type => E1
>> b-channel => 1:1-15
>> d-channel => 1:16
>> b-channel => 1:17-31
>>
>>
>>
>>
>> openzap.conf.xml
>>
>> <configuration name="openzap.conf" description="OpenZAP  
>> Configuration">
>> <settings>
>>   <param name="debug" value="7"/>
>>   <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>   <!--<param name="enable-analog-option" value="call-swap"/>-->
>>   <!--<param name="enable-analog-option" value="3-way"/>-->
>> </settings>
>>  <pri_spans>
>>    <span name="PRI_1">
>>      <!-- Log Levels: none, alert, crit, err, warning, notice, info,
>> debug -->
>>      <param name="q921loglevel" value="debug"/>
>>      <param name="q931loglevel" value="debug"/>
>>      <param name="mode" value="user"/>
>>      <param name="dialect" value="Oh this is not my dialect"/>
>>      <param name="dialplan" value="XML"/>
>>      <param name="context" value="default"/>
>>    </span>
>>  </pri_spans>
>> </configuration>
>>
>> Very interesting here is, that the "dialect" parameter doesn't seem  
>> to
>> have an effect on FS. I use that one above without any errors or  
>> warning
>> and I guess that was not intended.
>
> At this point in the OZ development we've got it set to default to
> "national" if the dialect isn't otherwise properly specified. It does
> make sense to throw an error if the dialect is not properly specified,
> even if we still default to national.
>
> -MC
>
>>
>>
>>
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.9 (MingW32)
>>
>> iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
>> BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
>> =geXp
>> -----END PGP SIGNATURE-----
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From astmac at stillnewt.org  Sat Dec 13 13:59:19 2008
From: astmac at stillnewt.org (Martin Joseph)
Date: Sat, 13 Dec 2008 13:59:19 -0800
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
	<830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
Message-ID: <52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>


On Dec 13, 2008, at 1:30 PM, Brian West wrote:

> I don't get what you're saying.. this looks 100% OK.  Also make sure
> you have the dev tools from apple and not automake, autoconf and
> libtool installed via fink,darwinports or any other such method.  If
> you can provide a bit more information maybe I can see what is wrong.
Yes, I think it looks good too.  But make fails with the Jira issue  
that has been going on for ages.

I have never used fink or darwinports or any other such methods on  
this box so that's out.

I definitely do have the Apple devtools for 10.4 installed.

I have no problem making the 1.01 FS from the tarball, but as it  
seems you are telling everyone to upgrade to the SVN trunk, I would  
love to do that also.  However, I am frustrated by my inability to  
get that going, as well as a severe lack of time.

Thanks for any help or ideas,
Marty


>
> /b
>
> On Dec 13, 2008, at 3:00 PM, martin joseph wrote:
>
>> I get the following and have issues making it...
>>
>> Updated to revision 10753.
>> mail:/usr/src/freeswitch/trunk root# ./bootstrap.sh
>> bootstrap: checking installation...
>> bootstrap: autoconf version 2.61 (ok)
>> bootstrap: automake version 1.10 (ok)
>> bootstrap: libtool version 1.5.24 (ok)
>>
>> Thanks for the info,
>> Marty
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From brian at freeswitch.org  Sat Dec 13 14:07:07 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Dec 2008 16:07:07 -0600
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
	<830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
	<52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>
Message-ID: <1848C35A-040F-4365-AA17-E43C9B32E11D@freeswitch.org>

Marty,
	Can you point out where its failing?  Nobody has been able to  
reproduce the issue that was reported on jira.  Even Anthony can't and  
he's on 10.4.  I'm on 10.5 and I don't have any issues either.  So if  
you can pin point the exact place where it fails we can look at it  
closer.

/b

On Dec 13, 2008, at 3:59 PM, Martin Joseph wrote:

> Yes, I think it looks good too.  But make fails with the Jira issue
> that has been going on for ages.
>
> I have never used fink or darwinports or any other such methods on
> this box so that's out.
>
> I definitely do have the Apple devtools for 10.4 installed.
>
> I have no problem making the 1.01 FS from the tarball, but as it
> seems you are telling everyone to upgrade to the SVN trunk, I would
> love to do that also.  However, I am frustrated by my inability to
> get that going, as well as a severe lack of time.
>
> Thanks for any help or ideas,
> Marty




From msc at freeswitch.org  Sat Dec 13 14:53:08 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 14:53:08 -0800
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <C0C90F5C-CC42-451A-AC91-D35BF4846EAE@jerris.com>
References: <49427CC6.2090407@ewetel.de>
	<87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>
	<C0C90F5C-CC42-451A-AC91-D35BF4846EAE@jerris.com>
Message-ID: <87f2f3b90812131453q7225c813s8c00a4388e082c40@mail.gmail.com>

Done: http://jira.freeswitch.org/browse/OPENZAP-37
-MC

On Sat, Dec 13, 2008 at 1:54 PM, Michael Jerris <mike at jerris.com> wrote:
> Please file a bug on this issue
>
> Mike
>
> On Dec 13, 2008, at 3:51 PM, "Michael Collins" <msc at freeswitch.org>
> wrote:
>
>> On Fri, Dec 12, 2008 at 7:01 AM, Helmut Kuper
>> <helmut.kuper at ewetel.de> wrote:
>>> -----BEGIN PGP SIGNED MESSAGE-----
>>> Hash: SHA1
>>>
>>> Hello,
>>>
>>> I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
>>> AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu
>>> 8.04
>>> server in a non-root environment.
>>>
>>> We experienced a timer problem which led to this FS console error
>>> message:
>>>
>>> [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries
>>>
>>>
>>> During anylizing this we found that q921 T203 is never reset when
>>> link
>>> is in state "Multiple Frame Mode Established" and SABME frames are
>>> received by FS. So it must timeout regardless if SABME frames are
>>> received or not.
>>> Additionally we found that the default T203 value (10 sec) was too
>>> short
>>> for AVAYA (it has to be >=19 sec)
>>>
>>> To fix the problem we changed two things in q921.c:
>>>
>>> Change T203 default value from 10 sec to 20000 sec
>>> Line 406: trunk->T203Timeout = 20000;
>>>
>>> Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
>>> received SABME frame
>>> Line 1996: Q921T203TimerReset(trunk, tei);
>>>
>>> After recompiling FS the Error disapeared. Next week we will do some
>>> calls over the link to make sure there are no other side effects.
>>>
>>> Is it planned to make the q921 timeouts configurable in
>>> openzap.conf or
>>> in openzap.conf.xml?
>>>
>>> best regards
>>> Helmut
>>>
>>>
>>> PS: My openzap configs:
>>>
>>> openzap.conf
>>>
>>> [span wanpipe PRI_1]
>>> trunk_type => E1
>>> b-channel => 1:1-15
>>> d-channel => 1:16
>>> b-channel => 1:17-31
>>>
>>>
>>>
>>>
>>> openzap.conf.xml
>>>
>>> <configuration name="openzap.conf" description="OpenZAP
>>> Configuration">
>>> <settings>
>>>   <param name="debug" value="7"/>
>>>   <!--<param name="hold-music" value="$${moh_uri}"/>-->
>>>   <!--<param name="enable-analog-option" value="call-swap"/>-->
>>>   <!--<param name="enable-analog-option" value="3-way"/>-->
>>> </settings>
>>>  <pri_spans>
>>>    <span name="PRI_1">
>>>      <!-- Log Levels: none, alert, crit, err, warning, notice, info,
>>> debug -->
>>>      <param name="q921loglevel" value="debug"/>
>>>      <param name="q931loglevel" value="debug"/>
>>>      <param name="mode" value="user"/>
>>>      <param name="dialect" value="Oh this is not my dialect"/>
>>>      <param name="dialplan" value="XML"/>
>>>      <param name="context" value="default"/>
>>>    </span>
>>>  </pri_spans>
>>> </configuration>
>>>
>>> Very interesting here is, that the "dialect" parameter doesn't seem
>>> to
>>> have an effect on FS. I use that one above without any errors or
>>> warning
>>> and I guess that was not intended.
>>
>> At this point in the OZ development we've got it set to default to
>> "national" if the dialect isn't otherwise properly specified. It does
>> make sense to throw an error if the dialect is not properly specified,
>> even if we still default to national.
>>
>> -MC
>>
>>>
>>>
>>>
>>> -----BEGIN PGP SIGNATURE-----
>>> Version: GnuPG v1.4.9 (MingW32)
>>>
>>> iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
>>> BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
>>> =geXp
>>> -----END PGP SIGNATURE-----
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From msc at freeswitch.org  Sat Dec 13 14:56:37 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 14:56:37 -0800
Subject: [Freeswitch-users] Error loading portaudio module
In-Reply-To: <661B71A1-D73F-43F0-97BA-11D6798D1486@jerris.com>
References: <20081212093859.GA7067@jdc.jasonjgw.net>
	<87f2f3b90812121158i3eb32681w2f1376a73af9efa@mail.gmail.com>
	<7b197bef0812121225o726bb0dfub48fbf1f25f1499@mail.gmail.com>
	<7b197bef0812121230x678ef817m3f135a303cb77fe2@mail.gmail.com>
	<20081212232403.GA7667@jdc.jasonjgw.net>
	<20081213005004.GA8393@jdc.jasonjgw.net>
	<20081213013538.GA16338@jdc.jasonjgw.net>
	<20081213020136.GA16597@jdc.jasonjgw.net>
	<661B71A1-D73F-43F0-97BA-11D6798D1486@jerris.com>
Message-ID: <87f2f3b90812131456s6bfe35eev8bbc1f04b831cdba@mail.gmail.com>

Done: http://jira.freeswitch.org/browse/FSBUILD-95

On Sat, Dec 13, 2008 at 1:51 PM, Michael Jerris <mike at jerris.com> wrote:
> Please file a bug on Jira.freeswitch.org on this issue so we make sure
> it gets adressef in the debs.
>
> Mike
>
> On Dec 12, 2008, at 9:01 PM, Jason White <jason at jasonjgw.net> wrote:
>
>> The problem is now solved.
>>
>> It turned out to be permissions: the freeswitch user wasn't added to
>> the audio
>> group in /etc/group, hence didn't have permission to interrogate the
>> audio
>> devices.
>>
>> Perhaps a future version of the Debian package could address this,
>> or at least
>> it should be noted somewhere.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From msc at freeswitch.org  Sat Dec 13 15:04:11 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 15:04:11 -0800
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <03c801c95d68$9880ce50$33014c0a@ws4>
References: <87f2f3b90812122239y127f35d6r80cd1286b7dddae8@mail.gmail.com>
	<03c801c95d68$9880ce50$33014c0a@ws4>
Message-ID: <87f2f3b90812131504ue83ddd3qa76a2379f013d531@mail.gmail.com>

On Sat, Dec 13, 2008 at 1:20 PM, Frank @ Impact <frank at impactfax.com> wrote:
> Michael,
>
> Got it working.  Just a little simpler then you outlined.
> I just added to my xml dialplan this line.
>
> <action application="set" data="execute_on_answer=lua uuid_send_dtmf.lua
> ${uuid} 20 123"/>
>
> I added this just before the bridge application.

Nice - simpler is almost always better! :)

>
> I did this instead of adding an extra extension to transfer to on
> answer.  Everything worked well.  The DTMF was played to the calling
> party.  Out of curiosity, if we wanted also to play the DTMF to the
> called party also, what would we have to give uuid_send_dtmf?
> Particularly since we call it before the bridge.
>

Definitely need the uuid of the leg in question. Could you pastebin or
email a sample dialplan? We could probably work it out together.

> Can uuid_send_dtmf accept anything like "w" for wait or anything else
> special for DTMF stuff?

The uuid_send_dtmf api cannot, but you could easily modify or create a
new version of my Lua script that accepts more (or different)
arguments. I suppose the trick there is that you'd need to read up on
Lua, which I highly recommend anyway because if you know Lua then you
can leverage some serious power in your dialplans.

>
> Also, I got an error output to the console when the sched_api was run.
> See below.
> ******
> 2008-12-13 16:07:28 [NOTICE] switch_cpp.cpp:1050 console_log() apicmd:
> sched_api
> 2008-12-13 16:07:28 [NOTICE] switch_cpp.cpp:1050 console_log() apiarg:
> +20 none uuid_send_dtmf 37618e54-c959-11dd-bc73-0923daa880b2 123
>
> 2008-12-13 16:07:28 [ERR] switch_cpp.cpp:1050 console_log() Result is
> +OK Added: 49751
> ******
> is this ERR anything to worry about even though we got a result ok?
>

I believe this "error" is innocuous. Sometimes the devs will log
certain events to the console as ERR so that they stand out during
debugging.
-MC
> Thanks again for the help.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Collins
>
> Frank,
>
> I found a simple way to handle this scenario. I decided just to create
> a small Lua script that would do the job. It's committed in latest
> trunk. Look in src/scripts/contrib/mcollins for uuid_send_dtmf.lua. It
> has comments on how to call it, including a sample dp call.
>
> The way I would use this in your scenario is to setup a destination
> using the execute_on_answer variable.
> http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer
>
> Have the destination be an extension that does something like this:
> <extension...>
>  <condition...>
>    <action application="answer"/>
>    <action application="lua" data="uuid_send_dtmf ${uuid} 10 123"/>
>    ...rest of diaplan...
>  </condition>
> </extension>
>
> The 10 means ten seconds, the 123 means send the dtmf digits 1,2,3 in
> order. You can tinker with the settings as you see fit.
>
> Please let me know how it goes. BTW, be sure to put the Lua script in
> /usr/local/freeswitch/scripts or specify the complete path name when
> calling the lua app in the dialplan.
>
> -MC
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From msc at freeswitch.org  Sat Dec 13 15:40:15 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 15:40:15 -0800
Subject: [Freeswitch-users] Maintaining call detail record
In-Reply-To: <614188.90296.qm@web30706.mail.mud.yahoo.com>
References: <614188.90296.qm@web30706.mail.mud.yahoo.com>
Message-ID: <87f2f3b90812131540v78272fe2m77ed194d07961200@mail.gmail.com>

Faisal,

A few things to keep in mind:

In cdr_csv.conf.xml you need to specify the correct template. There are
several templates specified. I don't know much about the perl script
mentioned there but I don't know if that is how I would approach the
situation personally. One of the templates is called "sql" and it creates
SQL statement for each cdr. You could literally pipe the contents of the cdr
file into MySQL and it will load the data into your table. However you will
need to handle the log file rotation. Some people use cron to send a HUP
signal to the freeswitch process which then rotates the log and Master.csv
files. I recommend you look at the cdr-csv directory before and after a
rotation so that you can see exactly what happens.

Here's a brief checklist for you to help you get going:
1 select the "sql" template in cdr_csv.conf.xml
 <param name="default-template" value="sql"/>

2 create a MySQL database for your cdr data

3 create a table called "cdr" (or rename the table used in the "sql"
template)
 the table needs to have all the fields laid out the way the template lays
them out
 this page can be used as a reference to get you started, but note the these
fields are NOT laid out the same way the sql template lays them out
 http://wiki.freeswitch.org/wiki/Mod_cdr#MySQL_Schema

4 decide how frequently you want to rotate log files and then set up a cron
job that sends the HUP signal:
 kill -hup `cat /usr/local/freeswitch/log/freeswitch.pid`
 After the kill -hup is sent your /usr/local/freeswitch/log/cdr-csv will
look something like this:
 -rw------- 1 root root        0 Dec 13 15:24 Master.csv
 -rw------- 1 root root  1473657 Dec 12 22:13 Master.csv.2008-12-13-15-24-16

  The file Master.csv.YYYY-MM-DD-hh-mm-ss now has the most recent CDR's.

5 run the most recent file through mysql. it is essentially just a text file
with a bunch of INSERT INTO statements
  mysql -u user -p password < Master.csv.YYYY-MM-DD-hh-mm-ss
  rm -f Master.csv.YYYY-MM-DD-hh-mm-ss

steps 4 and 5 could all be in the cron job which just has a script do all
the work. if you need assistance with setting up scripts and doing cron jobs
then i recommend that you manually do the steps one at a time and see
exactly what is happening and then learn how to do the shell script + cron
job.

Good luck!
-MC

P.S. - if anyone already has done all of this and is willing to share
his/her experiences please contact me off list as I would like to talk about
getting a wiki page set up that is dedicated to this sort of thing.

On Sat, Dec 13, 2008 at 1:44 AM, Faisal Maqsoodi <faisalmaqsoodi at yahoo.com>
wrote:
> How can i interface fs with mysql in order to maintain calls record like
> caller id and time n date of call etc. I ve worked on xml cdr but it
> contains too much info, more than i need and in a format which is not
easily
> understandable. I also tried perl coding mentioned on the link at the
bottom
> of the page
> http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r
> displayed during its execution. Is there any easy method for that. Plz
help
> me.
>

> faisal
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Dec 13 15:49:05 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 15:49:05 -0800
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers
In-Reply-To: <05FEA4243A6C422DB5A3D7838AA709FE@test>
References: <1229081927.4100.7.camel@gathern.lan>
	<05FEA4243A6C422DB5A3D7838AA709FE@test>
Message-ID: <87f2f3b90812131549g3ed2e25dg81ec4bea42d497fd@mail.gmail.com>

Also, the other question is this: do you *need* early media? If not then
Darren's suggestion is definitely the way to go. Note that if you ignore
early media then all calls that fail will show up as a NO ANSWER. If this
doesn't work for you then ignoring early media is not an option, in which
case there simply is no perfect way to do it and you just have to make the
best of it. What I've done in the past is something like this:

originate openzap/1/a/5551212 825551212

Then I define an extension that matches on ^82(\d+)$ and does something like
this
<action application="pre_answer"/>
<action application="set" data="execute_on_answer=transfer IVR_ANSWER"/>
<action application="sleep" data="20000"/>
...handle non-answered calls

Then I define another extension that matches on ^IVR_ANSWER$ and does
something like this
<action application="sleep" data="500"/>
<action application="ivr" data="my_ivr"/>
...etc...

The idea for me is to handle the different scenarios I might face when
dialing. At the very least if the call goes unanswered then I have the
hangup_cause variable that tells me if it was busy, no answer, invalid, etc.

Hope that helps.
-MC

On Fri, Dec 12, 2008 at 5:48 AM, Darren Schreiber <d at d-man.org> wrote:

> How are you originating calls? You probably need to add
> {ignore_early_media=true}. This tells FreeSWITCH not to return from
> origination when early media (progress/ringing) was received (I think
> anyway)...
>
> See http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media
>
> There is a sample of this in use with the originate command here:
> http://wiki.freeswitch.org/wiki/Mod_commands#originate (about halfway
> down)
>
> Setting channel variables before doing the originate
>
>   originate {ignore_early_media=true}sofia/
> mydomain.com/18005551212 at 1.2.3.4
> 15555551212
>
>
>
> Since you are making a dialer, you may want to start the originations in
> the
> background and move on to the next call while tweaking the timeout value
> for
> originated calls. From the WIKI again:
>
> "You can originate a call in the background (asynchronously) and playback a
> message with a 60 second timeout.
>
>   bgapi originate
> {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number
> &playback(message)"
>
> - Darren
>
>
>
> -----Original Message-----
> From: Alexandru Nedelcu [mailto:alex at sinapticode.ro]
> Sent: Friday, December 12, 2008 3:39 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Freeswitch streamFile when the user answers
>
> Hi,
>
> I'm working on a simple dialer, and I have the following problem: the audio
> file starts playing before the user answeres the phone (while it's
> ringing).
> It only works when I introduce a delay, but that doesn't seem right.
>
> For instance in the asterisk context referred in the call files, I had:
>
> exten => s,4,Answer
> exten => s,n,Wait(2)
> exten => s,n,Background(${SOUNDFILE})
> And indeed it played a soundfile 2 seconds after the called person picked
> up
> the phone
>
> In FS I currently initiate calls like this:
>
> session.waitForAnswer(10000);
>
> if (session.ready()) {
>    session.sleep(2000);
>    session.streamFile(/*...*/);
> }
>
> Is this right?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Sat Dec 13 15:52:59 2008
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 13 Dec 2008 15:52:59 -0800
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <op.ulwl5dbessieo1@woof.us.nortel.com>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
Message-ID: <87f2f3b90812131552p295eb1c4gee7d29f7a72624dc@mail.gmail.com>

Another option for you, ironically, is to have the freeswitch.log file plus
the other log files that are not freeswitch's, to go into a third directory
that is uniquely set up for this purpose. That way it wouldn't be disruptive
to move a bunch of files from log to db. All you'd have to do is modify the
logfile.conf.xml file and pick a new path for your freeswitch.log file...

-MC

On Tue, Dec 9, 2008 at 8:45 AM, Andy Spitzer <woof at nortel.com> wrote:

> Woof!
>
> It appears that FreeSWITCH writes
>
>   freeswitch.history
>   freeswitch.log
>   freeswitch.pid
>   freeswitch.xml.fsxml
>
> to the -log directory.
>
> Is there a way to put the files other than freeswitch.log into the -db
> directory instead?
>
> In my environment we archive and rotate everything in the log directory
> (which includes logs beside FreeSWITCH's), and these other FreeSWITCH files
> are getting rotated.  Yeah, I can explicitly exclude them, but to me it
> seems those really belong in the -db directory anyway, as they are
> inherently data needed for the current executable of FreeSWITCH, and not
> logs.
>
> --Woof!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From astmac at stillnewt.org  Sat Dec 13 16:23:44 2008
From: astmac at stillnewt.org (martin joseph)
Date: Sat, 13 Dec 2008 16:23:44 -0800
Subject: [Freeswitch-users] ./configure fails on 10.4.11
In-Reply-To: <1848C35A-040F-4365-AA17-E43C9B32E11D@freeswitch.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
	<830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
	<52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>
	<1848C35A-040F-4365-AA17-E43C9B32E11D@freeswitch.org>
Message-ID: <B9DF8866-2120-4E5C-8FFB-352A11A71438@stillnewt.org>


On Dec 13, 2008, at 2:07 PM, Brian West wrote:

> Marty,
> 	Can you point out where its failing?
./configure fails as follows (just as in the JIRA)

checking for a BSD-compatible install... /usr/bin/install -c
./configure: line 4112: syntax error near unexpected token  
`build_libtool_libs,'
./configure: line 4112: `    _LT_DECL(build_libtool_libs,  
enable_shared, 0,'
configure: error: ./configure.gnu failed for libs/openmrcp

> Nobody has been able to
> reproduce the issue that was reported on jira.  Even Anthony can't and
> he's on 10.4.
Huh,  I would love to figure this out, as it seems certain to be  
somehow specific to my install.
>   I'm on 10.5 and I don't have any issues either.  So if
> you can pin point the exact place where it fails we can look at it
> closer.
I would love that.

Thanks again for your excellent software and help.
Marty

>
> /b
>
> On Dec 13, 2008, at 3:59 PM, Martin Joseph wrote:
>
>> Yes, I think it looks good too.  But make fails with the Jira issue
>> that has been going on for ages.
>>
>> I have never used fink or darwinports or any other such methods on
>> this box so that's out.
>>
>> I definitely do have the Apple devtools for 10.4 installed.
>>
>> I have no problem making the 1.01 FS from the tarball, but as it
>> seems you are telling everyone to upgrade to the SVN trunk, I would
>> love to do that also.  However, I am frustrated by my inability to
>> get that going, as well as a severe lack of time.
>>
>> Thanks for any help or ideas,
>> Marty
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From chavpaskov at shaw.ca  Sat Dec 13 16:38:41 2008
From: chavpaskov at shaw.ca (Chav Paskov)
Date: Sat, 13 Dec 2008 16:38:41 -0800
Subject: [Freeswitch-users] how to handle  returned sip 302 dialplan
In-Reply-To: <9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
Message-ID: <49445591.90505@shaw.ca>

Brian West wrote:
> Chav,
> Once the 302 is received by FreeSWITCH it will follow it to the 
> contact listed in the 302.  What else are you needing to do?
>
> /b
>
> On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
>
>> *User-Agent: eXosip/3.1.0^M
>> Content-Length:  
>>
>>
>> my question is:
>>
>> Is it possible to send the call to z.z.z.z , receive the SIP 302 , 
>> process the data in Contact field  and redirect to the new destination 
>> contained in *Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
>> *without closing the session.
>> i red something about   <action application="set" 
>> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
>>
>> Any ideas on this matter  will be highly appreciated.
>> Best Regards
>> Chav
>>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
Thanks Brian,

probably i should have explained it in more details.
this whole thing started as an attempt to  implement lata ocn /local 
number portability/ instead of  pure per destination routing.
At the moment i have a access to  a service provider  who does 
"dipping"  and returns  the  LATA OCN  data associated with  any dialed 
destination number. it is returned as Contact:  and Content-length:  
fields in 302 message.

in other words:

1. i'm sending to this provider let say - 2025556666  as a destination 
number.
2. they do the dipping  and will return to me  either the new dest #  if 
2025556666 has been ported or if it was not
in content-length field they'll send  lata, ocn  and state and 10 digits 
number.
3. once received i have to  compare the received lata , ocn and state  
date with  a compiled  rate deck / blended from 5 different vendors/  
and pick the lowest rate - effectively building  LCR based on LATA OCN 
STATE info.

Hope this will help to clear the picture.
Regards
Chav





From frank at impactfax.com  Sat Dec 13 20:42:11 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 13 Dec 2008 23:42:11 -0500
Subject: [Freeswitch-users] schedule a DTMF tone into bridge
In-Reply-To: <87f2f3b90812131504ue83ddd3qa76a2379f013d531@mail.gmail.com>
Message-ID: <002701c95da6$54acb9d0$33014c0a@ws4>

Pretty simple actually...

<?xml version="1.0"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="Regex/XML Dialplan">
    <context name="default">
      <extension name="myextension">
        <condition>
	  <action application="enum" data="15555551212"/>
	  <action application="set" data="execute_on_answer=lua
uuid_send_dtmf.lua ${uuid} 10 55"/>
	  <!-- use this if to wrap just sections you want to test -->
	  <action application="tone_detect" data="test 941,1336 rw
+100000 hangup normal_clearing"/>
	  <action application="bridge" data="${enum_auto_route}"/> 
	  <action application="hangup"/>
        </condition>
      </extension>
    </context>
  </section>
</document>

BTW, this darn tone_detect is something I never could get working.  It
did not matter which side I sent the tone from, it never got trapped by
my test here.  The call never hung up on the tone, a 0.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Saturday, December 13, 2008 6:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] schedule a DTMF tone into bridge

>
> I did this instead of adding an extra extension to transfer to on
> answer.  Everything worked well.  The DTMF was played to the calling
> party.  Out of curiosity, if we wanted also to play the DTMF to the
> called party also, what would we have to give uuid_send_dtmf?
> Particularly since we call it before the bridge.
>

Definitely need the uuid of the leg in question. Could you pastebin or
email a sample dialplan? We could probably work it out together.






From jason at jasonjgw.net  Sat Dec 13 22:23:03 2008
From: jason at jasonjgw.net (Jason White)
Date: Sun, 14 Dec 2008 17:23:03 +1100
Subject: [Freeswitch-users] Sip profiles used in bridge application
Message-ID: <20081214062303.GA22331@jdc.jasonjgw.net>

The Wiki page at
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
gives the following example:

<action application="bridge"
data="sofia/sip/9998881111 at sip.yourprovider.com"/>

which, for me at least, doesn't work due to an invalid profile error:

[ERR] mod_sofia.c:2404 sofia_outgoing_channel() Invalid Profile

However, if I replace "sip" in the above action with "external" to specify the
external profile, it works.

I suspect this is a documentation error rather than a code bug. However, I
couldn't find an explanation of the syntax, in particular, whether the profile
name has to match a profile defined in the SIP configuration or whether there
are profile names defined in the code that have special meanings as well.

I'm using the default configuration for now.

I could go through the code to find this out, of course, but I thought it
better to ask here instead.




From msc at freeswitch.org  Sat Dec 13 22:39:55 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 13 Dec 2008 22:39:55 -0800
Subject: [Freeswitch-users] Sip profiles used in bridge application
In-Reply-To: <20081214062303.GA22331@jdc.jasonjgw.net>
References: <20081214062303.GA22331@jdc.jasonjgw.net>
Message-ID: <C1CD13F2-C731-4862-BFE7-0D71003592CC@freeswitch.org>

Jason,

Thanks for pointing this out. You are correct. This is a case of  
development moving faster than documentation efforts. I will update  
the wiki.

-MC

Sent from my iPhone

On Dec 13, 2008, at 10:23 PM, Jason White <jason at jasonjgw.net> wrote:

> The Wiki page at
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
> gives the following example:
>
> <action application="bridge"
> data="sofia/sip/9998881111 at sip.yourprovider.com"/>
>
> which, for me at least, doesn't work due to an invalid profile error:
>
> [ERR] mod_sofia.c:2404 sofia_outgoing_channel() Invalid Profile
>
> However, if I replace "sip" in the above action with "external" to  
> specify the
> external profile, it works.
>
> I suspect this is a documentation error rather than a code bug.  
> However, I
> couldn't find an explanation of the syntax, in particular, whether  
> the profile
> name has to match a profile defined in the SIP configuration or  
> whether there
> are profile names defined in the code that have special meanings as  
> well.
>
> I'm using the default configuration for now.
>
> I could go through the code to find this out, of course, but I  
> thought it
> better to ask here instead.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pabx_freeswitch at telenet.be  Sun Dec 14 04:52:17 2008
From: pabx_freeswitch at telenet.be (henkoegema)
Date: Sun, 14 Dec 2008 04:52:17 -0800 (PST)
Subject: [Freeswitch-users]  libtool version.
Message-ID: <21000027.post@talk.nabble.com>


root at MSI:/home/henkoegema/freeswitch# ./bootstrap.sh 
bootstrap: checking installation...
bootstrap: autoconf version 2.61 (ok)
bootstrap: automake version 1.10.1 (ok)
bootstrap: libtool version 2.2.4 found.    
           You need libtool version 1.5.14 or newer installed     
<------Isn't that what I have  :confused:
           to build FreeSWITCH from SVN.
root at MSI:/home/henkoegema/freeswitch# 

-- 
View this message in context: http://www.nabble.com/libtool-version.-tp21000027p21000027.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From jbr at consiglia.dk  Sun Dec 14 08:08:54 2008
From: jbr at consiglia.dk (Jon Bruel)
Date: Sun, 14 Dec 2008 17:08:54 +0100
Subject: [Freeswitch-users] How to control to domain used in INVITE From
	header
Message-ID: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE02838C515F@SBS2008SERVER.consiglia.local>

The situation is as follows: An incoming call is processed by the FS and sent out to a sip client. I want to control the From header in this outgoing INVITE.
I have tried to set various channel variables, including sip_h_From, in order to control to domain used in the INVITE From header, which for instance looks like this:
From: "JBS (Soft)" <sip:205 at X.X.X.X>
Instead of the server IP address, X.X.X.X, I want to set the SIP-domain used for the specific customer in a multi tenant setup. This from header is used by the phone telephone number list register (at least for the Snom phones), so controlling it is important. How is it done? /Jon
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From mike at jerris.com  Sun Dec 14 08:21:11 2008
From: mike at jerris.com (Michael Jerris)
Date: Sun, 14 Dec 2008 11:21:11 -0500
Subject: [Freeswitch-users] libtool version.
In-Reply-To: <21000027.post@talk.nabble.com>
References: <21000027.post@talk.nabble.com>
Message-ID: <08AE93ED-54E6-4179-BEEB-DF550DD05B5E@jerris.com>

http://jira.freeswitch.org/browse/FSBUILD-82

On Dec 14, 2008, at 7:52 AM, henkoegema wrote:

>
> root at MSI:/home/henkoegema/freeswitch# ./bootstrap.sh
> bootstrap: checking installation...
> bootstrap: autoconf version 2.61 (ok)
> bootstrap: automake version 1.10.1 (ok)
> bootstrap: libtool version 2.2.4 found.
>           You need libtool version 1.5.14 or newer installed
> <------Isn't that what I have  :confused:
>           to build FreeSWITCH from SVN.
> root at MSI:/home/henkoegema/freeswitch#
>
> -- 
> View this message in context: http://www.nabble.com/libtool-version.-tp21000027p21000027.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sun Dec 14 10:00:18 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Dec 2008 12:00:18 -0600
Subject: [Freeswitch-users] How to control to domain used in INVITE From
	header
In-Reply-To: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE02838C515F@SBS2008SERVER.consiglia.local>
References: <DE9AFF5DF0C8BB4881F86E50EAC0A9AE02838C515F@SBS2008SERVER.consiglia.local>
Message-ID: <704CEFC8-9E10-4690-AB8F-942D8382DF9E@freeswitch.org>

the sip_invite_domain variable.

/b

On Dec 14, 2008, at 10:08 AM, Jon Bruel wrote:

> The situation is as follows: An incoming call is processed by the FS  
> and sent out to a sip client. I want to control the >From header in  
> this outgoing INVITE.
>
> I have tried to set various channel variables, including sip_h_From,  
> in order to control to domain used in the INVITE From header, which  
> for instance looks like this:
>
> From: "JBS (Soft)" <sip:205 at X.X.X.X>
>
> Instead of the server IP address, X.X.X.X, I want to set the SIP- 
> domain used for the specific customer in a multi tenant setup. This  
> from header is used by the phone telephone number list register (at  
> least for the Snom phones), so controlling it is important. How is  
> it done? /Jon
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Dec 14 10:01:21 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Dec 2008 12:01:21 -0600
Subject: [Freeswitch-users] Sip profiles used in bridge application
In-Reply-To: <20081214062303.GA22331@jdc.jasonjgw.net>
References: <20081214062303.GA22331@jdc.jasonjgw.net>
Message-ID: <9B569932-48D8-4F66-9CDC-ECBEF7A6C459@freeswitch.org>

Well if the config had a profile called "sip" it would be perfectly  
fine.  Remember the profile names are not set in stone and you can  
name them what ever you wish.

/b

On Dec 14, 2008, at 12:23 AM, Jason White wrote:

>
> [ERR] mod_sofia.c:2404 sofia_outgoing_channel() Invalid Profile




From jason at jasonjgw.net  Sun Dec 14 14:45:35 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 09:45:35 +1100
Subject: [Freeswitch-users] references to source files in error messages
Message-ID: <20081214224535.GA5335@jdc.jasonjgw.net>

I would just like to thank the FreeSWITCH developers for including the source
file names and line numbers in error messages. This is not only helpful to the
authors of the software, but to anyone who can read C code.




From jason at jasonjgw.net  Sun Dec 14 20:41:13 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 15:41:13 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
Message-ID: <20081215044113.GA9555@jdc.jasonjgw.net>

I'm trying to make an outbound call to an IPv6 host using the pre-supplied
internal-ipv6 profile.

<action application="bridge"
data="sofia/internal-ipv6/nnnn at ipv6-host.domain"/>

I can ping the host with ping6, and it has a DNS AAAA record. (There is no A
record as the host is a friend's box behind a NAT).

My own box is also behind a NAT, but it has IPv6 connectivity via a tunnel
broker.

The error in the freeswitch.log file is:
[ERR] sofia_reg.c:1344 sofia_reg_handle_sip_r_challenge() No Matching gateway
found

It's quite possible I could be doing something wrong, or there might be a bug
somewhere, or the provided internal-ipv6 profile might need some adjusting...

Suggestions welcome.




From jason at jasonjgw.net  Sun Dec 14 21:25:18 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 16:25:18 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215044113.GA9555@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
Message-ID: <20081215052518.GA9967@jdc.jasonjgw.net>

Incidentally, if I try it with the external profile, or with my own
external-ipv6 profile (the same as the supplied one, but binding to IPv6
rather than IPv4 addresses), then I get

2008-12-15 16:16:50 [DEBUG] mod_sofia.c:253 sofia_on_hangup()
sofia/external-ipv6/nnnn at ipv6-host.domain Overriding SIP cause 503 with 503
from the other leg

The other party (with a configured FreeSWITCH listening at the IPv6 address in
question) gets the same error upon attempting to call my machine.




From faisalmaqsoodi at yahoo.com  Sun Dec 14 21:43:12 2008
From: faisalmaqsoodi at yahoo.com (Faisal Maqsoodi)
Date: Sun, 14 Dec 2008 21:43:12 -0800 (PST)
Subject: [Freeswitch-users] Maintaining call detail record
Message-ID: <627709.35367.qm@web30703.mail.mud.yahoo.com>

                        
                        

if (typeof YAHOO == "undefined") {
 var YAHOO = {};
}
YAHOO.Shortcuts = YAHOO.Shortcuts || {};
YAHOO.Shortcuts.hasSensitiveText = false;
YAHOO.Shortcuts.sensitivityType = [];
YAHOO.Shortcuts.doUlt = false;
YAHOO.Shortcuts.location = "us";
YAHOO.Shortcuts.document_id = 0;
YAHOO.Shortcuts.document_type = "";
YAHOO.Shortcuts.document_title = "[Freeswitch-users] Maintaining call detail record";
YAHOO.Shortcuts.document_publish_date = "";
YAHOO.Shortcuts.document_author = "faisalmaqsoodi at yahoo.com";
YAHOO.Shortcuts.document_url = "";
YAHOO.Shortcuts.document_tags = "";
YAHOO.Shortcuts.document_language = "english";
YAHOO.Shortcuts.annotationSet = {
"lw_1229319648_0": {
"text": "caller id",
"extended": 0,
"startchar": 188,
"endchar": 196,
"start": 188,
"end": 196,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 0.2054,
"relScore": 5.33095,
"type": ["shortcuts:/concept"],
"category": ["CONCEPT"],
"wikiId": "Caller_ID",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked",
"metaData": {
"visible": "true"
}
 },
"lw_1229319648_1": {
"text": "http://wiki.freeswitch.org/wiki/Mod_cdr_csv,",
"extended": 0,
"startchar": 426,
"endchar": 469,
"start": 426,
"end": 469,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 1,
"relScore": 0,
"type": ["shortcuts:/us/instance/identifier/URL"],
"category": ["IDENTIFIER"],
"wikiId": "",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "",
"metaData": {
"visible": "true"
}
 }
};
YAHOO.Shortcuts.headerID = "284c8f98b4fb0aebc968053934caa66b";

How can i interface fs with mysql in order to maintain calls record like caller id
and time n date of call etc. I ve worked on xml cdr but it contains too
much info, more than i need and in a format which is not easily
understandable. I also tried perl coding mentioned on the link at the
bottom of the page
http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r displayed during its execution. Is there any easy method for that. Plz help me.
 
?????????????????????????????????????????????????????????????????????????????????????? faisal



      
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From helmut.kuper at ewetel.de  Sun Dec 14 23:24:57 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 15 Dec 2008 08:24:57 +0100
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <87f2f3b90812131453q7225c813s8c00a4388e082c40@mail.gmail.com>
References: <49427CC6.2090407@ewetel.de>	<87f2f3b90812131251u1be13350k271179147291be2e@mail.gmail.com>	<C0C90F5C-CC42-451A-AC91-D35BF4846EAE@jerris.com>
	<87f2f3b90812131453q7225c813s8c00a4388e082c40@mail.gmail.com>
Message-ID: <49460649.6030302@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1


Hello Michael 1+2,

thank you for opening a bug. Otherwise I would have opened a bug today
after testing it a bit more.

regards
Helmut

Am 13.12.2008 23:53, schrieb Michael Collins:
> Done: http://jira.freeswitch.org/browse/OPENZAP-37
> -MC
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RdcAnRasalavYRJ9hRcj5DjWYooZS6vb
=cPz3
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From helmut.kuper at ewetel.de  Sun Dec 14 23:35:17 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 15 Dec 2008 08:35:17 +0100
Subject: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue
In-Reply-To: <191c3a030812120718n7d8c5410y2ad3cebab8f5be3b@mail.gmail.com>
References: <49427CC6.2090407@ewetel.de>
	<191c3a030812120718n7d8c5410y2ad3cebab8f5be3b@mail.gmail.com>
Message-ID: <494608B5.4060305@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

OK, I opened a bug for T203 on jira:
http://jira.freeswitch.org/browse/OPENZAP-38

regards
helmut

Am 12.12.2008 16:18, schrieb Anthony Minessale:
> if you open a jira issue on it we can probably add your patch and/or the
> config option.
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59QAoKlallmRwdKBuTOUJcVZMDgQL0bU
=idOI
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From stevecrozz at gmail.com  Sun Dec 14 23:32:08 2008
From: stevecrozz at gmail.com (Stephen Crosby)
Date: Sun, 14 Dec 2008 23:32:08 -0800
Subject: [Freeswitch-users] running custom script with bind_meta_app
Message-ID: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>

I'm Stephen Crosby, and I've just started working with freeswitch.
It's been great so far.

I want to run a custom script inside a conference when a DTMF sequence
is entered. I found bind_meta_app and thought it would be perfect, but
I can't seem to get it to work. When I dial-in and press *8, I get no
debugging output at all. When I press another sequence like *9 for
instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf()
sofia/external/5593495805 at sip.gafachi.com Ignoring meta digit '9' not
mapped. The script I wrote has been tested with "jsrun script.js" from
the command line and it does work. I've got the debugging level all
the way up and there's just not much for me to go on. Any help would
be greatly appreciated.

<extension name="gafachi-DID1">
  <condition field="destination_number" expression="^xxxxxxxxxxx$">
    <action application="answer"/>
    <action application="bind_meta_app" data="8 ab s
javascript::$${base_dir}/scripts/script.js"/>
    <action application="conference" data="3000 at default"/>
  </condition>
</extension>

--Stephen



From jason at jasonjgw.net  Mon Dec 15 00:01:05 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 19:01:05 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215052518.GA9967@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
Message-ID: <20081215080105.GA11352@jdc.jasonjgw.net>

Turning on all of the Sofia debugging options reveals the following.

FS sends out the invite (UDP, 1245 bytes).

Then we get:

nta: timer shortened to 500 ms
tport_wakeup_pri(0x2490700): events ERR
tport_udp_error: icmp(6) message was truncated (at 832)
tport_udp_error: Message too long (90) [icmp6 type=2 code=0 info=00000500]
	reported by [2001:470:0:5d::2]:65535
tport_release(0x2490700): 0x7fe244098c10 by 0x25022a0 with (nil)
nta: INVITE (108544297): Message too long (90) with udp/[xxxxx - remote host's
IPv6 address]:5060
nua(0x7fe24409af90): event r_invite 503 Service Unavailable
nua(0x7fe24409af90): call state changed: calling -> init
nua: nua_application_event: entering
nua(0x7fe24409af90): event i_state 503 Service Unavailable
nua(0x7fe24409af90): event i_terminated 503 Service Unavailable
nua: nua_handle_magic: entering
nua(0x7fe24409af90): removing session usage
nua: nua_application_event: entering
nta_leg_destroy(0x7fe2440822f0)
2008-12-15 18:14:27 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/external-ipv6/nnnn at remote-host.domain entering state [terminated]

The IPv6 address that sent the "message too long" error is presumably that of
a router between my FreeSWITCH box and my friend's FreeSWITCH box.

Is there a way around this?




From jan.kubr at gmail.com  Mon Dec 15 01:27:25 2008
From: jan.kubr at gmail.com (Jan Kubr)
Date: Mon, 15 Dec 2008 10:27:25 +0100
Subject: [Freeswitch-users] Interrupting read application with DTMF
Message-ID: <698401620812150127s4f96a7c4h3516da007f7399a0@mail.gmail.com>

Hi,
I have been having some troubles with the read application for quite a
while which I haven't been able to solve yet.
I have Freeswitch connected to a SIP gateway to accept calls from a
landline-like number. For the incoming calls I have a simple testing
dialplan:

<action application="read" data="1 1
en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
<action application="phrase" data="spell,${res}"/>

The behavior I have a problem with is that the read app should
terminate when I press a digit and the execution should jump to the
next action - meaning the playback of the file should be interrupted.
The problem is that when I call the public number from my cell phone
this works only about 50% of the time. In the other cases I need to
wait for the wav file to be played (or press the digit two or three
times). When using a SIP phone it always works.

Today I tried to convert the wav file the read app plays to the GSM
format and found out it fixed the problem! Now I can almost always
interrupt the read app with DTMF from my cell phone. Doing the same
from my SIP phone doesn't work well though when the file is GSM.

Can someone explain me what is going on here and what is the right
approach? I'm on revision 10751. I've tried to set a few configuration
variables based on suggestions from this list, but it didn't make any
difference.

Thanks,
Jan Kubr



From jonas.gauffin at gmail.com  Mon Dec 15 02:19:44 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 15 Dec 2008 11:19:44 +0100
Subject: [Freeswitch-users] Bridging through gateway
Message-ID: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>

I'm trying to bridge using a non-registered gateway. And I
get MANDATORY_IE_MISSING back. Why is that?

2008-12-15 11:06:49 [DEBUG] switch_core_state_machine.c:152
switch_core_standard_on_execute()
sofia/internal/u1000044 at 192.168.1.112:5070Execute
bridge(sofia/default/
0236661201 at sip-corporate2.tele2.se)
[.....]
2008-12-15 11:06:49 [DEBUG] sofia.c:2511 sofia_handle_sip_i_state() Channel
sofia/internal/0236661201 at sip-corporate2.tele2.se entering state [calling]
2008-12-15 11:06:49 [ERR] sofia_reg.c:1312
sofia_reg_handle_sip_r_challenge() No Matching gateway found
2008-12-15 11:06:49 [NOTICE] sofia_reg.c:1333
sofia_reg_handle_sip_r_challenge() Hangup sofia/internal/
0236661201 at sip-corporate2.tele2.se [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING]
2008-12-15 11:06:49 [DEBUG] switch_channel.c:1478
switch_channel_perform_hangup() Send signal sofia/internal/
0236661201 at sip-corporate2.tele2.se [KILL]
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From FranziskaRoehler at aol.com  Mon Dec 15 01:57:57 2008
From: FranziskaRoehler at aol.com (=?iso-8859-1?Q?Franziska_R=F6hler?=)
Date: Mon, 15 Dec 2008 10:57:57 +0100
Subject: [Freeswitch-users] Openzap ERROR can't dial
Message-ID: <000c01c95e9b$9cbaf610$6445310a@Franzi>

Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

 

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN). But when I make an outbound call, I have an error. FS
can?t dial.

 

[ERR] zap_isdn.c:559 state_advance() 1:1 STATE [DIALING]

 

I have this warnings too, when no call is done:

 

2008-12-15 10:11:14 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6

2008-12-15 10:11:15 [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 7

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:1
(1:1) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:2
(1:2) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:3
(1:3) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:4
(1:4) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:5
(1:5) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:6
(1:6) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:7
(1:7) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:8
(1:8) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:9
(1:9) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:10
(1:10) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:11
(1:11) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:12
(1:12) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:13
(1:13) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:14
(1:14) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:15
(1:15) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:16
(1:17) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:17
(1:18) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:18
(1:19) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:19
(1:20) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:20
(1:21) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:21
(1:22) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:22
(1:23) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:23
(1:24) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:24
(1:25) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:25
(1:26) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:26
(1:27) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:27
(1:28) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:28
(1:29) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:29
(1:30) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:30
(1:31) has alarms [YELLOW]

2008-12-15 10:11:15 [WARNING] zap_isdn.c:803 process_event() channel 1:31
(1:16) has alarms [YELLOW]

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:1

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:2

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:3

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:4

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:5

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:6

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:7

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:8

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:9

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:10

2008-12-15 10:11:20 [WARNING] mod_openzap.c:1405 on_clear_channel_signal()
Unhandled type for channel 1:11

 

What do I wrong?

I hope you can help me!

 

Here my configuration:

 

OPENZAP.CONF

 

[span zt]

name => OpenZap

number => 1

trunk_type => e1

b-channel => 1-15, 17-31

d-channel => 16

 

OPENZAP.CONF.XML

 

<configuration name="openzap.conf" description="OpenZAP Configuration">

  <settings>

    <param name="debug" value="0"/>

  </settings>

  <pri_spans>

    <span id="1">

      <param name="mode" value="user"/>

      <param name="dialect" value="euro"/>

      <param name="dialplan" value="XML"/>

      <param name="context" value="default"/>

    </span>

  </pri_spans>

</configuration>

 

ZT.CONF

 

[defaults]

codec_ms => 20

wink_ms => 150

flash_ms => 750

echo_cancel_level => 64

 

DEFAULT.XML

 

   <extension name="outgoing_pri">

      <condition field="destination_number"
expression="^([0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9])$">

            <action application="set" data="dialed_ext=$1"/>

            <action application="bridge" data="openzap/1/A/${dialed_ext}"/>

      </condition>

    </extension>

 

When I have forget to display some Configuration to , you can tell me!
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From jason at jasonjgw.net  Mon Dec 15 03:03:46 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Dec 2008 22:03:46 +1100
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
Message-ID: <20081215110346.GA12681@jdc.jasonjgw.net>

On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
> I'm trying to bridge using a non-registered gateway. And I
> get MANDATORY_IE_MISSING back. Why is that?

Does the gateway allow unauthenticated clients to make calls? If you obtain a
SIP trace, you'll be able to see whether it's an authentication issue.

As an aside, it would be an improvement to FreeSWITCH if Sofia debugging could
be turned on and off within a running FreeSWITCH instance, including SIP
traces, instead of the administrator's having to restart FreeSWITCH with
environment variables exported, as is presently required according to the
wiki.




From brian at freeswitch.org  Mon Dec 15 06:45:49 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 08:45:49 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215080105.GA11352@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
Message-ID: <38E75A1D-7391-46A8-BD9C-1C851A019625@freeswitch.org>

Are you using SVN trunk?  This has been fixed already as far as I  
remember!!

/b

Sent from my iPhne

On Dec 15, 2008, at 2:01 AM, Jason White <jason at jasonjgw.net> wrote:

> Turning on all of the Sofia debugging options reveals the following.
>
> FS sends out the invite (UDP, 1245 bytes).
>
> Then we get:
>
> nta: timer shortened to 500 ms
> tport_wakeup_pri(0x2490700): events ERR
> tport_udp_error: icmp(6) message was truncated (at 832)
> tport_udp_error: Message too long (90) [icmp6 type=2 code=0  
> info=00000500]
>    reported by [2001:470:0:5d::2]:65535
> tport_release(0x2490700): 0x7fe244098c10 by 0x25022a0 with (nil)
> nta: INVITE (108544297): Message too long (90) with udp/[xxxxx -  
> remote host's
> IPv6 address]:5060
> nua(0x7fe24409af90): event r_invite 503 Service Unavailable
> nua(0x7fe24409af90): call state changed: calling -> init
> nua: nua_application_event: entering
> nua(0x7fe24409af90): event i_state 503 Service Unavailable
> nua(0x7fe24409af90): event i_terminated 503 Service Unavailable
> nua: nua_handle_magic: entering
> nua(0x7fe24409af90): removing session usage
> nua: nua_application_event: entering
> nta_leg_destroy(0x7fe2440822f0)
> 2008-12-15 18:14:27 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state()  
> Channel sofia/external-ipv6/nnnn at remote-host.domain entering state  
> [terminated]
>
> The IPv6 address that sent the "message too long" error is  
> presumably that of
> a router between my FreeSWITCH box and my friend's FreeSWITCH box.
>
> Is there a way around this?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Dec 15 07:05:49 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 09:05:49 -0600
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <20081215110346.GA12681@jdc.jasonjgw.net>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
	<20081215110346.GA12681@jdc.jasonjgw.net>
Message-ID: <432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>

If you don't need auth you don't need a gateway.  sofia/profile/ 
number at remoteip is all you should need.

/b

On Dec 15, 2008, at 5:03 AM, Jason White wrote:

> On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
>> I'm trying to bridge using a non-registered gateway. And I
>> get MANDATORY_IE_MISSING back. Why is that?
>
> Does the gateway allow unauthenticated clients to make calls? If you  
> obtain a
> SIP trace, you'll be able to see whether it's an authentication issue.
>
> As an aside, it would be an improvement to FreeSWITCH if Sofia  
> debugging could
> be turned on and off within a running FreeSWITCH instance, including  
> SIP
> traces, instead of the administrator's having to restart FreeSWITCH  
> with
> environment variables exported, as is presently required according  
> to the
> wiki.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Mon Dec 15 07:08:56 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 09:08:56 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215080105.GA11352@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
Message-ID: <081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>

 From the past emails and the data you have provided so far it makes  
me believe you're not on SVN trunk.  Also the second email looks like  
the far end tries to challenge you and we can't find a matching  
gateway.   I have tested IPv6 with Snom and FS to FS pretty much  
daily.   Can you verify you're on SVN trunk and not 1.0.1?

/b

On Dec 15, 2008, at 2:01 AM, Jason White wrote:

> Turning on all of the Sofia debugging options reveals the following.
>
> FS sends out the invite (UDP, 1245 bytes).
>
> Then we get:
>
> nta: timer shortened to 500 ms
> tport_wakeup_pri(0x2490700): events ERR
> tport_udp_error: icmp(6) message was truncated (at 832)
> tport_udp_error: Message too long (90) [icmp6 type=2 code=0  
> info=00000500]
> 	reported by [2001:470:0:5d::2]:65535
> tport_release(0x2490700): 0x7fe244098c10 by 0x25022a0 with (nil)
> nta: INVITE (108544297): Message too long (90) with udp/[xxxxx -  
> remote host's
> IPv6 address]:5060
> nua(0x7fe24409af90): event r_invite 503 Service Unavailable
> nua(0x7fe24409af90): call state changed: calling -> init
> nua: nua_application_event: entering
> nua(0x7fe24409af90): event i_state 503 Service Unavailable
> nua(0x7fe24409af90): event i_terminated 503 Service Unavailable
> nua: nua_handle_magic: entering
> nua(0x7fe24409af90): removing session usage
> nua: nua_application_event: entering
> nta_leg_destroy(0x7fe2440822f0)
> 2008-12-15 18:14:27 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state()  
> Channel sofia/external-ipv6/nnnn at remote-host.domain entering state  
> [terminated]
>
> The IPv6 address that sent the "message too long" error is  
> presumably that of
> a router between my FreeSWITCH box and my friend's FreeSWITCH box.
>
> Is there a way around this?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Mon Dec 15 07:09:13 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 09:09:13 -0600
Subject: [Freeswitch-users] running custom script with bind_meta_app
In-Reply-To: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>
References: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>
Message-ID: <910AFB7A-9CDE-49AE-A72D-7942F8F79DA3@freeswitch.org>

What are you wanting to accomplish first?

/b

On Dec 15, 2008, at 1:32 AM, Stephen Crosby wrote:

> I'm Stephen Crosby, and I've just started working with freeswitch.
> It's been great so far.
>
> I want to run a custom script inside a conference when a DTMF sequence
> is entered. I found bind_meta_app and thought it would be perfect, but
> I can't seem to get it to work. When I dial-in and press *8, I get no
> debugging output at all. When I press another sequence like *9 for
> instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf()
> sofia/external/5593495805 at sip.gafachi.com Ignoring meta digit '9' not
> mapped. The script I wrote has been tested with "jsrun script.js" from
> the command line and it does work. I've got the debugging level all
> the way up and there's just not much for me to go on. Any help would
> be greatly appreciated.
>
> <extension name="gafachi-DID1">
>  <condition field="destination_number" expression="^xxxxxxxxxxx$">
>    <action application="answer"/>
>    <action application="bind_meta_app" data="8 ab s
> javascript::$${base_dir}/scripts/script.js"/>
>    <action application="conference" data="3000 at default"/>
>  </condition>
> </extension>
>
> --Stephen

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From freeswitch at ptmm.com  Mon Dec 15 02:56:12 2008
From: freeswitch at ptmm.com (Clifford)
Date: Mon, 15 Dec 2008 03:56:12 -0700
Subject: [Freeswitch-users] Multiple mod_portaudio channels
Message-ID: <mailman.9.1269220377.3324.freeswitch-users@lists.freeswitch.org>

I need to set up a system that would allow multiple mod_portaudio channels
to be configured.  
 
The calls would come in over SIP and any calls would be assigned a specific
port on the multi-port sound card.
 
I would probably set it up so specific extensions go to specific ports, for
example 1001 would be port 1 and 1008 would be port 8 on the sound card.
 
The documentation on the mod_portaudio module is very slim.  In fact I would
say it is nearly undocumented.  Is it capable of handling multiple audio
device ports (such as an 8-port LYNX sound card)? 
 
If so how would it be interfaced/configured?
 
Thanks,
 
Clifford 
 
 
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From saeedahmad1981 at gmail.com  Mon Dec 15 07:25:43 2008
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 15 Dec 2008 16:25:43 +0100
Subject: [Freeswitch-users] Newbie Questions
Message-ID: <293ED6E2D87647248E526DFA0C08462C@SaeedLaptop>

Hi,

I am very new Freeswitch.

Till now I've some experience with Asterisk.

Can someone explain me the following things:

1. Can I connect my TDM switch to Freeswitch? My switch can speak Dss1/ss7,
SS7 is more preferable 
2. Is freeswitch similar to nextone SBC?
(http://www.nextpointnetworks.com/)
3. Does freeswtich support codec translation?

These are very basic questions at start. When I'll go deeper into it then
there could be more questions.

Kind Regards
Saeed




From jflowers at ezo.net  Mon Dec 15 07:45:38 2008
From: jflowers at ezo.net (jflowers)
Date: Mon, 15 Dec 2008 07:45:38 -0800 (PST)
Subject: [Freeswitch-users]  Speed Dial Emulation
Message-ID: <21016167.post@talk.nabble.com>


How do I emulate a speed dial setup.  That is, from extension 1003 I dial
just a 1 ( or 2, or 3 etc.) and nothing else and freeswitch dials a PSTN
number.  Is there software to do this?


-- 
View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21016167.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From anthony.minessale at gmail.com  Mon Dec 15 07:48:08 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Dec 2008 09:48:08 -0600
Subject: [Freeswitch-users] Interrupting read application with DTMF
In-Reply-To: <698401620812150127s4f96a7c4h3516da007f7399a0@mail.gmail.com>
References: <698401620812150127s4f96a7c4h3516da007f7399a0@mail.gmail.com>
Message-ID: <191c3a030812150748o60877e95yf071302c37d55b4@mail.gmail.com>

I think your solution is most likely superstition and that your real problem
is related
to your cell phone and the PSTN to SIP translation somewhere along the way.

I bet if you called in to the same extension with a SIP desk phone, that it
would work every
time no matter what format your file is.


On Mon, Dec 15, 2008 at 3:27 AM, Jan Kubr <jan.kubr at gmail.com> wrote:

> Hi,
> I have been having some troubles with the read application for quite a
> while which I haven't been able to solve yet.
> I have Freeswitch connected to a SIP gateway to accept calls from a
> landline-like number. For the incoming calls I have a simple testing
> dialplan:
>
> <action application="read" data="1 1
> en/us/callie/conference/8000/conf-pin.wav res 10000 #"/>
> <action application="phrase" data="spell,${res}"/>
>
> The behavior I have a problem with is that the read app should
> terminate when I press a digit and the execution should jump to the
> next action - meaning the playback of the file should be interrupted.
> The problem is that when I call the public number from my cell phone
> this works only about 50% of the time. In the other cases I need to
> wait for the wav file to be played (or press the digit two or three
> times). When using a SIP phone it always works.
>
> Today I tried to convert the wav file the read app plays to the GSM
> format and found out it fixed the problem! Now I can almost always
> interrupt the read app with DTMF from my cell phone. Doing the same
> from my SIP phone doesn't work well though when the file is GSM.
>
> Can someone explain me what is going on here and what is the right
> approach? I'm on revision 10751. I've tried to set a few configuration
> variables based on suggestions from this list, but it didn't make any
> difference.
>
> Thanks,
> Jan Kubr
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From wasim at convergence.pk  Mon Dec 15 07:49:19 2008
From: wasim at convergence.pk (Wasim Baig)
Date: Mon, 15 Dec 2008 20:49:19 +0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <293ED6E2D87647248E526DFA0C08462C@SaeedLaptop>
References: <293ED6E2D87647248E526DFA0C08462C@SaeedLaptop>
Message-ID: <b8ad2a5b0812150749t75e3c058u77f05fc5a91b5bbe@mail.gmail.com>

On Mon, Dec 15, 2008 at 8:25 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

Hi,


Salaam Saeed.


> I am very new Freeswitch.


Welcome.


> Till now I've some experience with Asterisk.


Be prepared to be amazed.

Can someone explain me the following things:
>
> 1. Can I connect my TDM switch to Freeswitch? My switch can speak Dss1/ss7,
> SS7 is more preferable


Currently, there is no open source ss7 implementation for FS. You can use
Sangoma's SMG with FS though.


> 2. Is freeswitch similar to nextone SBC?
> (http://www.nextpointnetworks.com/)


It can act as an SBC. See http://wiki.freeswitch.org/wiki/Specsheet
for more details.

3. Does freeswtich support codec translation?


Yes, it does for the supported codecs http://wiki.freeswitch.org/wiki/Codecs


> These are very basic questions at start. When I'll go deeper into it then
> there could be more questions.


Do read up at http://wiki.freeswitch.org/wiki/Main_Page

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as
you scope creep, so shall we reap ...
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From gmaruzz at celliax.org  Mon Dec 15 07:58:35 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 15 Dec 2008 16:58:35 +0100
Subject: [Freeswitch-users] Multiple mod_portaudio channels
In-Reply-To: <49467c03.030bca0a.2a43.ffff90cfSMTPIN_ADDED@mx.google.com>
References: <49467c03.030bca0a.2a43.ffff90cfSMTPIN_ADDED@mx.google.com>
Message-ID: <7b197bef0812150758r611e44c3re692902213891da6@mail.gmail.com>

You can try making the lynx appear to Operating System like 8 single
soundcards (mono 1in 1out).
You can do this on Linux via the /etc/asound.conf

Then for each soundcard you created, you can start one portaudio channel


Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Mon, Dec 15, 2008 at 11:56 AM, Clifford <freeswitch at ptmm.com> wrote:
> I need to set up a system that would allow multiple mod_portaudio channels
> to be configured.
>
> The calls would come in over SIP and any calls would be assigned a specific
> port on the multi-port sound card.
>
> I would probably set it up so specific extensions go to specific ports, for
> example 1001 would be port 1 and 1008 would be port 8 on the sound card.
>
> The documentation on the mod_portaudio module is very slim.  In fact I would
> say it is nearly undocumented.  Is it capable of handling multiple audio
> device ports (such as an 8-port LYNX sound card)?
>
> If so how would it be interfaced/configured?
>
> Thanks,
>
> Clifford
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jonas.gauffin at gmail.com  Mon Dec 15 08:01:01 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Mon, 15 Dec 2008 17:01:01 +0100
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
	<20081215110346.GA12681@jdc.jasonjgw.net>
	<432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>
Message-ID: <d021d3be0812150801g3d904fb5xe98b281f142a2d95@mail.gmail.com>

Yeah I know. That's what I'm trying to do, but with the domain name (
sip-corporate2.tele2.se) instead of the  ip.
I'm not sure that the gateway works without authentication and I'm wondering
if MANDATORY_IE_MISSING means that the gateway wants authentication or if it
means something else.
It's the "No Matching gateway found" message that is confusing, since I'm
not trying to use a registered gateway?

On Mon, Dec 15, 2008 at 4:05 PM, Brian West <brian at freeswitch.org> wrote:

> If you don't need auth you don't need a gateway.  sofia/profile/
> number at remoteip is all you should need.
>
> /b
>
> On Dec 15, 2008, at 5:03 AM, Jason White wrote:
>
> > On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
> >> I'm trying to bridge using a non-registered gateway. And I
> >> get MANDATORY_IE_MISSING back. Why is that?
> >
> > Does the gateway allow unauthenticated clients to make calls? If you
> > obtain a
> > SIP trace, you'll be able to see whether it's an authentication issue.
> >
> > As an aside, it would be an improvement to FreeSWITCH if Sofia
> > debugging could
> > be turned on and off within a running FreeSWITCH instance, including
> > SIP
> > traces, instead of the administrator's having to restart FreeSWITCH
> > with
> > environment variables exported, as is presently required according
> > to the
> > wiki.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From woof at nortel.com  Mon Dec 15 08:03:56 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 15 Dec 2008 11:03:56 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <87f2f3b90812131552p295eb1c4gee7d29f7a72624dc@mail.gmail.com>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
	<87f2f3b90812131552p295eb1c4gee7d29f7a72624dc@mail.gmail.com>
Message-ID: <op.ul7n8uh5ssieo1@woof.us.nortel.com>

Woof!

On Sat, 13 Dec 2008 18:52:59 -0500, Michael Collins <msc at freeswitch.org> wrote:

> All you'd have to do is modify the logfile.conf.xml file and pick a new path for your freeswitch.log file...

I agree.  I had discovered this option and considered it as a workaround.  Then I also found that mod_xml_rpc was also logging in log dir, and I haven't found a way to control that one--I haven't looked that hard, I must admit.  Weekends are a great way to forget everything you were doing the week before!


--Woof!



From rjcajax at gmail.com  Mon Dec 15 08:05:52 2008
From: rjcajax at gmail.com (Robert Clayton)
Date: Mon, 15 Dec 2008 11:05:52 -0500
Subject: [Freeswitch-users] Recording Pause
Message-ID: <d7e909fc0812150805k1f166ce4sdf4699c7a395f7e1@mail.gmail.com>

All,

I was thinking since there is no direct functionality for the person being
recorded to pause the recording could this be done indirectly.
For example if using the functionality to record only when voice is present
could the audio stream be preprocessed to evaluate the  dtmf where if a
specific key was pressed the audio stream would no longer be passed or
passed as blank to the recording functionality so an implicit pause would be
created. And when the person recording wished to continue pressing a key
would signal FS to pass the audio again therefore reestablishing recording
to the same file?

Bob



From chavpaskov at shaw.ca  Mon Dec 15 08:15:38 2008
From: chavpaskov at shaw.ca (Chav Paskov)
Date: Mon, 15 Dec 2008 08:15:38 -0800
Subject: [Freeswitch-users] how to handle  returned sip 302 dialplan
In-Reply-To: <49445591.90505@shaw.ca>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
	<49445591.90505@shaw.ca>
Message-ID: <494682AA.9090200@shaw.ca>

Chav Paskov wrote:
> Brian West wrote:
>   
>> Chav,
>> Once the 302 is received by FreeSWITCH it will follow it to the 
>> contact listed in the 302.  What else are you needing to do?
>>
>> /b
>>
>> On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
>>
>>     
>>> *User-Agent: eXosip/3.1.0^M
>>> Content-Length:  
>>>
>>>
>>> my question is:
>>>
>>> Is it possible to send the call to z.z.z.z , receive the SIP 302 , 
>>> process the data in Contact field  and redirect to the new destination 
>>> contained in *Contact: <sip:2145551234 at 64.112.188.84>;npdi^M
>>> *without closing the session.
>>> i red something about   <action application="set" 
>>> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
>>>
>>> Any ideas on this matter  will be highly appreciated.
>>> Best Regards
>>> Chav
>>>
>>>       
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
> Thanks Brian,
>
> probably i should have explained it in more details.
> this whole thing started as an attempt to  implement lata ocn /local 
> number portability/ instead of  pure per destination routing.
> At the moment i have a access to  a service provider  who does 
> "dipping"  and returns  the  LATA OCN  data associated with  any dialed 
> destination number. it is returned as Contact:  and Content-length:  
> fields in 302 message.
>
> in other words:
>
> 1. i'm sending to this provider let say - 2025556666  as a destination 
> number.
> 2. they do the dipping  and will return to me  either the new dest #  if 
> 2025556666 has been ported or if it was not
> in content-length field they'll send  lata, ocn  and state and 10 digits 
> number.
> 3. once received i have to  compare the received lata , ocn and state  
> date with  a compiled  rate deck / blended from 5 different vendors/  
> and pick the lowest rate - effectively building  LCR based on LATA OCN 
> STATE info.
>
> Hope this will help to clear the picture.
> Regards
> Chav
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   




From sicfslist at gmail.com  Mon Dec 15 08:40:13 2008
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Mon, 15 Dec 2008 10:40:13 -0600
Subject: [Freeswitch-users] how to handle returned sip 302 dialplan
In-Reply-To: <49445591.90505@shaw.ca>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
	<49445591.90505@shaw.ca>
Message-ID: <35b355e90812150840k362096cah292647c92b8681a0@mail.gmail.com>

Chav,
We recently / are still going through the same process (in order to route on
LRN) vs NPANXX or LATA based routing.  Here was the best way that we came up
with to do it:
  -- we use xml_curl exclusively for routing decisions
  -- so in the cgi script that xml_curl hits one of the things it can (and
does based on certain parameters) is fire off a url to another LNP server
that we built
  -- the LNP server actually does the dip (either from a cache) and returns
the info

We felt this was much better for a few reasons:
  -- caching the LNP data for a 24 hour period would save us in excess of
$100k a year
  -- having a specialized mechanism to do this was much easier to implement
for the cgi process than supporting 302 redirects directly on the FS boxes
was much easier (which just wasn't possible with the cgi mechanism)
  -- every LNP provider returns 302's slightly different ... so we didn't
want to have to reinvent the wheel on the FS machines if we ever wanted to
add redundancy or switch providers

Guess it all depends on your config ... but this was the easiest and most
cost-effective means for us to implement.


On Sat, Dec 13, 2008 at 6:38 PM, Chav Paskov <chavpaskov at shaw.ca> wrote:

> Brian West wrote:
> > Chav,
> > Once the 302 is received by FreeSWITCH it will follow it to the
> > contact listed in the 302.  What else are you needing to do?
> >
> > /b
> >
> > On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
> >
> >> *User-Agent: eXosip/3.1.0^M
> >> Content-Length:
> >>
> >>
> >> my question is:
> >>
> >> Is it possible to send the call to z.z.z.z , receive the SIP 302 ,
> >> process the data in Contact field  and redirect to the new destination
> >> contained in *Contact: <sip:2145551234 at 64.112.188.84<sip%3A2145551234 at 64.112.188.84>
> >;npdi^M
> >> *without closing the session.
> >> i red something about   <action application="set"
> >> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
> >>
> >> Any ideas on this matter  will be highly appreciated.
> >> Best Regards
> >> Chav
> >>
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> Thanks Brian,
>
> probably i should have explained it in more details.
> this whole thing started as an attempt to  implement lata ocn /local
> number portability/ instead of  pure per destination routing.
> At the moment i have a access to  a service provider  who does
> "dipping"  and returns  the  LATA OCN  data associated with  any dialed
> destination number. it is returned as Contact:  and Content-length:
> fields in 302 message.
>
> in other words:
>
> 1. i'm sending to this provider let say - 2025556666  as a destination
> number.
> 2. they do the dipping  and will return to me  either the new dest #  if
> 2025556666 has been ported or if it was not
> in content-length field they'll send  lata, ocn  and state and 10 digits
> number.
> 3. once received i have to  compare the received lata , ocn and state
> date with  a compiled  rate deck / blended from 5 different vendors/
> and pick the lowest rate - effectively building  LCR based on LATA OCN
> STATE info.
>
> Hope this will help to clear the picture.
> Regards
> Chav
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saeedahmad1981 at gmail.com  Mon Dec 15 08:42:25 2008
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 15 Dec 2008 17:42:25 +0100
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <b8ad2a5b0812150749t75e3c058u77f05fc5a91b5bbe@mail.gmail.com>
References: <293ED6E2D87647248E526DFA0C08462C@SaeedLaptop>
	<b8ad2a5b0812150749t75e3c058u77f05fc5a91b5bbe@mail.gmail.com>
Message-ID: <FB4BFC95627B448DA2627A2321DE6C9C@SaeedLaptop>

Wsalam Waseem sb.

Good to see you on mailing list :-)

 

So i hope you already know why I am trying to move to FS, I am still trying
to do similar thing, blind transfer of zap call with number changed (22)
release cause with new number, since I'll do it with freeswitch do you think
its possible now?

- Saeed

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wasim
Baig
Sent: Monday, December 15, 2008 4:49 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie Questions

 

On Mon, Dec 15, 2008 at 8:25 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>
wrote:

Hi,


Salaam Saeed.
 

I am very new Freeswitch.


Welcome.
 

Till now I've some experience with Asterisk.


Be prepared to be amazed. 

Can someone explain me the following things:

1. Can I connect my TDM switch to Freeswitch? My switch can speak Dss1/ss7,
SS7 is more preferable


Currently, there is no open source ss7 implementation for FS. You can use
Sangoma's SMG with FS though.
 

2. Is freeswitch similar to nextone SBC?
(http://www.nextpointnetworks.com/)


It can act as an SBC. See http://wiki.freeswitch.org/wiki/Specsheet
for more details.

3. Does freeswtich support codec translation?


Yes, it does for the supported codecs http://wiki.freeswitch.org/wiki/Codecs
 

These are very basic questions at start. When I'll go deeper into it then
there could be more questions.


Do read up at http://wiki.freeswitch.org/wiki/Main_Page 


-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as
you scope creep, so shall we reap ... 

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From chavpaskov at shaw.ca  Mon Dec 15 08:45:22 2008
From: chavpaskov at shaw.ca (Chav Paskov)
Date: Mon, 15 Dec 2008 08:45:22 -0800
Subject: [Freeswitch-users] how to handle returned sip 302 dialplan
In-Reply-To: <35b355e90812150840k362096cah292647c92b8681a0@mail.gmail.com>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
	<49445591.90505@shaw.ca>
	<35b355e90812150840k362096cah292647c92b8681a0@mail.gmail.com>
Message-ID: <494689A2.7080303@shaw.ca>

Shelby Ramsey wrote:
> Chav,
>
> We recently / are still going through the same process (in order to 
> route on LRN) vs NPANXX or LATA based routing.  Here was the best way 
> that we came up with to do it:
>   -- we use xml_curl exclusively for routing decisions
>   -- so in the cgi script that xml_curl hits one of the things it can 
> (and does based on certain parameters) is fire off a url to another 
> LNP server that we built
>   -- the LNP server actually does the dip (either from a cache) and 
> returns the info
>
> We felt this was much better for a few reasons:
>   -- caching the LNP data for a 24 hour period would save us in excess 
> of $100k a year
>   -- having a specialized mechanism to do this was much easier to 
> implement for the cgi process than supporting 302 redirects directly 
> on the FS boxes was much easier (which just wasn't possible with the 
> cgi mechanism)
>   -- every LNP provider returns 302's slightly different ... so we 
> didn't want to have to reinvent the wheel on the FS machines if we 
> ever wanted to add redundancy or switch providers
>
> Guess it all depends on your config ... but this was the easiest and 
> most cost-effective means for us to implement.
>
>
> On Sat, Dec 13, 2008 at 6:38 PM, Chav Paskov <chavpaskov at shaw.ca 
> <mailto:chavpaskov at shaw.ca>> wrote:
>
>     Brian West wrote:
>     > Chav,
>     > Once the 302 is received by FreeSWITCH it will follow it to the
>     > contact listed in the 302.  What else are you needing to do?
>     >
>     > /b
>     >
>     > On Dec 13, 2008, at 2:24 PM, Chav Paskov wrote:
>     >
>     >> *User-Agent: eXosip/3.1.0^M
>     >> Content-Length:
>     >>
>     >>
>     >> my question is:
>     >>
>     >> Is it possible to send the call to z.z.z.z , receive the SIP 302 ,
>     >> process the data in Contact field  and redirect to the new
>     destination
>     >> contained in *Contact: <sip:2145551234 at 64.112.188.84
>     <mailto:sip%3A2145551234 at 64.112.188.84>>;npdi^M
>     >> *without closing the session.
>     >> i red something about   <action application="set"
>     >> data="continue_on_fail=true"/>  but i'm not sure  how to use it.
>     >>
>     >> Any ideas on this matter  will be highly appreciated.
>     >> Best Regards
>     >> Chav
>     >>
>     >
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > Freeswitch-users mailing list
>     > Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     Thanks Brian,
>
>     probably i should have explained it in more details.
>     this whole thing started as an attempt to  implement lata ocn /local
>     number portability/ instead of  pure per destination routing.
>     At the moment i have a access to  a service provider  who does
>     "dipping"  and returns  the  LATA OCN  data associated with  any
>     dialed
>     destination number. it is returned as Contact:  and Content-length:
>     fields in 302 message.
>
>     in other words:
>
>     1. i'm sending to this provider let say - 2025556666  as a destination
>     number.
>     2. they do the dipping  and will return to me  either the new dest
>     #  if
>     2025556666 has been ported or if it was not
>     in content-length field they'll send  lata, ocn  and state and 10
>     digits
>     number.
>     3. once received i have to  compare the received lata , ocn and state
>     date with  a compiled  rate deck / blended from 5 different vendors/
>     and pick the lowest rate - effectively building  LCR based on LATA OCN
>     STATE info.
>
>     Hope this will help to clear the picture.
>     Regards
>     Chav
>
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
thanks for the prompt response.
Can you pls  give me an example how to access the info  contained in  
Contact:  and content-legth: fields  if you can.
I was thinking  in going the exactly same direction  in terms of 
building  xml_curl  dialplan  but i'm lacking knowledge
on how to access variables.
Regards
Chav



From intralanman at freeswitch.org  Mon Dec 15 08:53:49 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Mon, 15 Dec 2008 11:53:49 -0500
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <d021d3be0812150801g3d904fb5xe98b281f142a2d95@mail.gmail.com>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>	<20081215110346.GA12681@jdc.jasonjgw.net>	<432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>
	<d021d3be0812150801g3d904fb5xe98b281f142a2d95@mail.gmail.com>
Message-ID: <49468B9D.5010903@freeswitch.org>

posting relevant pieces of your dialplan and sofia configs would 
probably help a bit.

-Ray


Jonas Gauffin wrote:
> Yeah I know. That's what I'm trying to do, but with the domain name 
> (sip-corporate2.tele2.se <http://sip-corporate2.tele2.se>) instead of 
> the  ip.
>
> I'm not sure that the gateway works without authentication and I'm 
> wondering if MANDATORY_IE_MISSING means that the gateway wants 
> authentication or if it means something else.
> It's the "No Matching gateway found" message that is confusing, since 
> I'm not trying to use a registered gateway?
>
> On Mon, Dec 15, 2008 at 4:05 PM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     If you don't need auth you don't need a gateway.  sofia/profile/
>     number at remoteip is all you should need.
>
>     /b
>
>     On Dec 15, 2008, at 5:03 AM, Jason White wrote:
>
>     > On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
>     >> I'm trying to bridge using a non-registered gateway. And I
>     >> get MANDATORY_IE_MISSING back. Why is that?
>     >
>     > Does the gateway allow unauthenticated clients to make calls? If you
>     > obtain a
>     > SIP trace, you'll be able to see whether it's an authentication
>     issue.
>     >
>     > As an aside, it would be an improvement to FreeSWITCH if Sofia
>     > debugging could
>     > be turned on and off within a running FreeSWITCH instance, including
>     > SIP
>     > traces, instead of the administrator's having to restart FreeSWITCH
>     > with
>     > environment variables exported, as is presently required according
>     > to the
>     > wiki.
>     >
>     >
>     > _______________________________________________
>     > Freeswitch-users mailing list
>     > Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From brian at freeswitch.org  Mon Dec 15 08:55:10 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 10:55:10 -0600
Subject: [Freeswitch-users] how to handle returned sip 302 dialplan
In-Reply-To: <494689A2.7080303@shaw.ca>
References: <494419F9.6090304@shaw.ca>
	<9A6B5C53-5827-4261-91F8-DC5624C4F505@freeswitch.org>
	<49445591.90505@shaw.ca>
	<35b355e90812150840k362096cah292647c92b8681a0@mail.gmail.com>
	<494689A2.7080303@shaw.ca>
Message-ID: <97AF9163-4EE6-43D6-A232-A0534519F723@freeswitch.org>

I fear that you won't be able to get at any information in a 302  
issued to FreeSWITCH as those are on auto pilot.

You can try toying with the info application to see if it gets at the  
info you need.

/b

On Dec 15, 2008, at 10:45 AM, Chav Paskov wrote:

> thanks for the prompt response.
> Can you pls  give me an example how to access the info  contained in
> Contact:  and content-legth: fields  if you can.
> I was thinking  in going the exactly same direction  in terms of
> building  xml_curl  dialplan  but i'm lacking knowledge
> on how to access variables.
> Regards
> Chav




From brian at freeswitch.org  Mon Dec 15 08:56:53 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 10:56:53 -0600
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <op.ulwl5dbessieo1@woof.us.nortel.com>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
Message-ID: <F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>

I can't figure out why the log file would need to be in the db folder...

/b

On Dec 9, 2008, at 10:45 AM, Andy Spitzer wrote:

> Woof!
>
> It appears that FreeSWITCH writes
>
>   freeswitch.history
>   freeswitch.log
>   freeswitch.pid
>   freeswitch.xml.fsxml
>
> to the -log directory.
>
> Is there a way to put the files other than freeswitch.log into the - 
> db directory instead?
>
> In my environment we archive and rotate everything in the log  
> directory (which includes logs beside FreeSWITCH's), and these other  
> FreeSWITCH files are getting rotated.  Yeah, I can explicitly  
> exclude them, but to me it seems those really belong in the -db  
> directory anyway, as they are inherently data needed for the current  
> executable of FreeSWITCH, and not logs.
>
> --Woof!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From stevecrozz at gmail.com  Mon Dec 15 09:25:58 2008
From: stevecrozz at gmail.com (Stephen Crosby)
Date: Mon, 15 Dec 2008 09:25:58 -0800
Subject: [Freeswitch-users] running custom script with bind_meta_app
In-Reply-To: <910AFB7A-9CDE-49AE-A72D-7942F8F79DA3@freeswitch.org>
References: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>
	<910AFB7A-9CDE-49AE-A72D-7942F8F79DA3@freeswitch.org>
Message-ID: <11990ade0812150925i33e9cf6ex6a1fe53c423fae9b@mail.gmail.com>

I just want to listen for some DTMF sequence while in a conference.
The conference host should be able to enter the sequence at any time
(and any number of times) to run a custom script. I've already written
one in javascript, but I can rewrite it in another language if it's
easier.

On Mon, Dec 15, 2008 at 7:09 AM, Brian West <brian at freeswitch.org> wrote:
> What are you wanting to accomplish first?
> /b
> On Dec 15, 2008, at 1:32 AM, Stephen Crosby wrote:
>
> I'm Stephen Crosby, and I've just started working with freeswitch.
> It's been great so far.
>
> I want to run a custom script inside a conference when a DTMF sequence
> is entered. I found bind_meta_app and thought it would be perfect, but
> I can't seem to get it to work. When I dial-in and press *8, I get no
> debugging output at all. When I press another sequence like *9 for
> instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf()
> sofia/external/5593495805 at sip.gafachi.com Ignoring meta digit '9' not
> mapped. The script I wrote has been tested with "jsrun script.js" from
> the command line and it does work. I've got the debugging level all
> the way up and there's just not much for me to go on. Any help would
> be greatly appreciated.
>
> <extension name="gafachi-DID1">
>  <condition field="destination_number" expression="^xxxxxxxxxxx$">
>    <action application="answer"/>
>    <action application="bind_meta_app" data="8 ab s
> javascript::$${base_dir}/scripts/script.js"/>
>    <action application="conference" data="3000 at default"/>
>  </condition>
> </extension>
>
> --Stephen
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From woof at nortel.com  Mon Dec 15 09:35:24 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 15 Dec 2008 12:35:24 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
	<F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>
Message-ID: <op.ul7shal3ssieo1@woof.us.nortel.com>

Woof!

On Mon, 15 Dec 2008 11:56:53 -0500, Brian West <brian at freeswitch.org> wrote:

> I can't figure out why the log file would need to be in the db folder...

I think you misunderstand.

It's these files:
   freeswitch.history
   freeswitch.pid
   freeswitch.xml.fsxml

That I feel would be better off in the db folder.  They are not logs, and should not be rotated.

--Woof!



From intralanman at freeswitch.org  Mon Dec 15 10:16:32 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Mon, 15 Dec 2008 13:16:32 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <op.ul7shal3ssieo1@woof.us.nortel.com>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>	<F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>
	<op.ul7shal3ssieo1@woof.us.nortel.com>
Message-ID: <49469F00.7020203@freeswitch.org>


> I think you misunderstand.
>
> It's these files:
>    freeswitch.history
>    freeswitch.pid
>    freeswitch.xml.fsxml
>
> That I feel would be better off in the db folder.  They are not logs, and should not be rotated.
>
>   


if freeswitch.history isn't a log, what is it? seems to me taht it's a 
log of what commands you've run recently... it's definitely NOT a 
database.... neither is the pid file. while the pid file isn't a db, 
it's also not really a log... but i don't know that i'd agree with 
making a "run" directory just to house the pid.

-Ray



From kkielhofner at star2star.com  Mon Dec 15 10:35:18 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Mon, 15 Dec 2008 13:35:18 -0500
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215080105.GA11352@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
Message-ID: <2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>

On Mon, Dec 15, 2008 at 3:01 AM, Jason White <jason at jasonjgw.net> wrote:
> Turning on all of the Sofia debugging options reveals the following.
>
> FS sends out the invite (UDP, 1245 bytes).
>
> Then we get:
>
> nta: timer shortened to 500 ms
> tport_wakeup_pri(0x2490700): events ERR
> tport_udp_error: icmp(6) message was truncated (at 832)
> tport_udp_error: Message too long (90) [icmp6 type=2 code=0 info=00000500]
>        reported by [2001:470:0:5d::2]:65535
> tport_release(0x2490700): 0x7fe244098c10 by 0x25022a0 with (nil)
> nta: INVITE (108544297): Message too long (90) with udp/[xxxxx - remote host's
> IPv6 address]:5060
> nua(0x7fe24409af90): event r_invite 503 Service Unavailable
> nua(0x7fe24409af90): call state changed: calling -> init
> nua: nua_application_event: entering
> nua(0x7fe24409af90): event i_state 503 Service Unavailable
> nua(0x7fe24409af90): event i_terminated 503 Service Unavailable
> nua: nua_handle_magic: entering
> nua(0x7fe24409af90): removing session usage
> nua: nua_application_event: entering
> nta_leg_destroy(0x7fe2440822f0)
> 2008-12-15 18:14:27 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/external-ipv6/nnnn at remote-host.domain entering state [terminated]
>
> The IPv6 address that sent the "message too long" error is presumably that of
> a router between my FreeSWITCH box and my friend's FreeSWITCH box.
>
> Is there a way around this?

What happens if you use TCP transport instead of UDP?  You're probably
running into the infamous SIP UDP fragmentation (there isn't any)
problem.


-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From woof at nortel.com  Mon Dec 15 10:37:15 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 15 Dec 2008 13:37:15 -0500
Subject: [Freeswitch-users] Where FreeSWITCH writes some files
In-Reply-To: <49469F00.7020203@freeswitch.org>
References: <op.ulwl5dbessieo1@woof.us.nortel.com>
	<F0D9B4FD-FDAE-46B7-A0AD-3EB08E48A78F@freeswitch.org>
	<op.ul7shal3ssieo1@woof.us.nortel.com>
	<49469F00.7020203@freeswitch.org>
Message-ID: <op.ul7vcd14ssieo1@woof.us.nortel.com>

Woof!

On Mon, 15 Dec 2008 13:16:32 -0500, Raymond Chandler <intralanman at freeswitch.org> wrote:

> if freeswitch.history isn't a log, what is it? seems to me taht it's a
> log of what commands you've run recently... it's definitely NOT a
> database....

Actually, I the readline/history library uses it to determine the command line history (http://tiswww.case.edu/php/chet/readline/history.html#SEC15).  So it IS a database.  It may also be seen as a log of commands, as it happens to be a nice ASCII file, but that's an intended side effect of the way the library writes it.


--Woof!



From brian at freeswitch.org  Mon Dec 15 10:43:08 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 12:43:08 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>
Message-ID: <4D744169-0A93-44C7-AEDF-89C78F279AE2@freeswitch.org>

Well its important to know if he's on 1.0.1 or SVN trunk.  In 1.0.1  
the retry via TCP was disabled on ipv6 and that has now been  
corrected.  If it gets the message too big error it'll turn around and  
requeue the invite on TCP without any user interaction.

/b

On Dec 15, 2008, at 12:35 PM, Kristian Kielhofner wrote:

>
> What happens if you use TCP transport instead of UDP?  You're probably
> running into the infamous SIP UDP fragmentation (there isn't any)
> problem.




From brian at freeswitch.org  Mon Dec 15 11:46:07 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 13:46:07 -0600
Subject: [Freeswitch-users] ./configure fails on 10.4.11
In-Reply-To: <B9DF8866-2120-4E5C-8FFB-352A11A71438@stillnewt.org>
References: <6299C8F9-5080-41F3-AE96-2F79068E7A87@stillnewt.org>
	<830E4144-4649-4117-9CB7-466145A2A79E@freeswitch.org>
	<52908180-AB47-41F7-BA2F-CBFCD9DF6A16@stillnewt.org>
	<1848C35A-040F-4365-AA17-E43C9B32E11D@freeswitch.org>
	<B9DF8866-2120-4E5C-8FFB-352A11A71438@stillnewt.org>
Message-ID: <A9592378-9B8D-41D1-82D1-1ACE1CF3E165@freeswitch.org>

Anyway you can get one of us onto that OS X machine?  We haven't been  
able to reproduce this and without access to a machine its taking  
place on we can't fix it.

/b

On Dec 13, 2008, at 6:23 PM, martin joseph wrote:

>
> On Dec 13, 2008, at 2:07 PM, Brian West wrote:
>
>> Marty,
>> 	Can you point out where its failing?
> ./configure fails as follows (just as in the JIRA)
>
> checking for a BSD-compatible install... /usr/bin/install -c
> ./configure: line 4112: syntax error near unexpected token
> `build_libtool_libs,'
> ./configure: line 4112: `    _LT_DECL(build_libtool_libs,
> enable_shared, 0,'
> configure: error: ./configure.gnu failed for libs/openmrcp
>
>> Nobody has been able to
>> reproduce the issue that was reported on jira.  Even Anthony can't  
>> and
>> he's on 10.4.
> Huh,  I would love to figure this out, as it seems certain to be
> somehow specific to my install.
>>  I'm on 10.5 and I don't have any issues either.  So if
>> you can pin point the exact place where it fails we can look at it
>> closer.
> I would love that.
>
> Thanks again for your excellent software and help.
> Marty
>
>>
>> /b
>>
>> On Dec 13, 2008, at 3:59 PM, Martin Joseph wrote:
>>
>>> Yes, I think it looks good too.  But make fails with the Jira issue
>>> that has been going on for ages.
>>>
>>> I have never used fink or darwinports or any other such methods on
>>> this box so that's out.
>>>
>>> I definitely do have the Apple devtools for 10.4 installed.
>>>
>>> I have no problem making the 1.01 FS from the tarball, but as it
>>> seems you are telling everyone to upgrade to the SVN trunk, I would
>>> love to do that also.  However, I am frustrated by my inability to
>>> get that going, as well as a severe lack of time.
>>>
>>> Thanks for any help or ideas,
>>> Marty
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From kkielhofner at star2star.com  Mon Dec 15 12:06:39 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Mon, 15 Dec 2008 15:06:39 -0500
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <4D744169-0A93-44C7-AEDF-89C78F279AE2@freeswitch.org>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>
	<4D744169-0A93-44C7-AEDF-89C78F279AE2@freeswitch.org>
Message-ID: <2d9149cd0812151206y8df6d3cn8cb998e914698019@mail.gmail.com>

On Mon, Dec 15, 2008 at 1:43 PM, Brian West <brian at freeswitch.org> wrote:
> Well its important to know if he's on 1.0.1 or SVN trunk.  In 1.0.1
> the retry via TCP was disabled on ipv6 and that has now been
> corrected.  If it gets the message too big error it'll turn around and
> requeue the invite on TCP without any user interaction.
>
> /b
>

...which is exactly what the RFC says to do (IPv6 or IPv4).

Bravo!

I thought (for the user) it might make sense to just force TCP.  It
might make sense to set TCP explicitly (less back and forth).
However, keep in mind the higher overhead of TCP in the first place.
I guess it would depend on what's in the body (obviously).  Messages
with smaller bodies (SDPs, etc) might eek through with UDP while those
with larger bodies may have to requeue for TCP.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From brian at freeswitch.org  Mon Dec 15 12:14:50 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 14:14:50 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <2d9149cd0812151206y8df6d3cn8cb998e914698019@mail.gmail.com>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<2d9149cd0812151035k33055ca6x78a9e66dee58290f@mail.gmail.com>
	<4D744169-0A93-44C7-AEDF-89C78F279AE2@freeswitch.org>
	<2d9149cd0812151206y8df6d3cn8cb998e914698019@mail.gmail.com>
Message-ID: <114ADAD5-3964-431B-8E47-D8B2FD8E7393@freeswitch.org>

Pretty much everything in ipv6 will be over the MTU...  on ipv4 we  
disabled this because when you talk to things that don't speak TCP you  
have a packet over the MTU you try TCP it times out in 30 seconds then  
you send it anyway via UDP.  So why have a 30 seconds timeout when you  
don't need too.

/b

On Dec 15, 2008, at 2:06 PM, Kristian Kielhofner wrote:

> I thought (for the user) it might make sense to just force TCP.  It
> might make sense to set TCP explicitly (less back and forth).
> However, keep in mind the higher overhead of TCP in the first place.
> I guess it would depend on what's in the body (obviously).  Messages
> with smaller bodies (SDPs, etc) might eek through with UDP while those
> with larger bodies may have to requeue for TCP.




From jason at jasonjgw.net  Mon Dec 15 14:54:51 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Dec 2008 09:54:51 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
Message-ID: <20081215225451.GA5592@jdc.jasonjgw.net>

On Mon, Dec 15, 2008 at 09:08:56AM -0600, Brian West wrote:
>  From the past emails and the data you have provided so far it makes  
> me believe you're not on SVN trunk.  

I'm on trunk as of December 12, revision 10725. If it was fixed since then, I
can easily recompile.

The other party is on a recent revision from trunk and also runs into the same
problems in trying to call me via IPv6.




From brian at freeswitch.org  Mon Dec 15 15:09:21 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 17:09:21 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215225451.GA5592@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
	<20081215225451.GA5592@jdc.jasonjgw.net>
Message-ID: <0DEA56F4-9C15-47FA-8746-924CD6EF4109@freeswitch.org>

Well I just tested this again with the latest svn trunk

d77374e2-a381-4eb1-8771-5ccdae2155fd,2008-12-15 17:05:51,1229382351,sofia/internal/1000 at bkw.org 
,CS_EXECUTE,Brian West,1000,99.157.44.200,9888,bridge,sofia/internal- 
ipv6/888@[2001:470:7:ea::2],XML,default,G722,16000,G722,16000
0ed0b7ee-5de5-44a3-9f96-f6bbc5a7e0ba,2008-12-15  
17:05:51,1229382351,sofia/internal-ipv6/888@[2001:470:7:ea:: 
2],CS_EXCHANGE_MEDIA,Brian West,1000,99.157.44.200,888@[2001:470:7:ea:: 
2],,,XML,default,G722,16000,G722,16000

   1.447392 2001:470:1f0f:142:21a:92ff:fe3f:6a0f -> 2001:470:7:ea::2  
RTP PT=ITU-T G.722, SSRC=0x89B50B10, Seq=64494, Time=2387361962
   1.460530 2001:470:7:ea::2 -> 2001:470:1f0f:142:21a:92ff:fe3f:6a0f  
RTP PT=ITU-T G.722, SSRC=0xEB44C067, Seq=37131, Time=1328480

Seems to work fine for me... care to let me at your machine to take a  
look?

Works great did an ipv4 call to my FS box and ipv6 out to the  
conference box.


Try this out

sofia/internal-ipv6/888@[2001:470:7:ea::2]

(correct the profile for your needs)

/b

On Dec 15, 2008, at 4:54 PM, Jason White wrote:

> On Mon, Dec 15, 2008 at 09:08:56AM -0600, Brian West wrote:
>> From the past emails and the data you have provided so far it makes
>> me believe you're not on SVN trunk.
>
> I'm on trunk as of December 12, revision 10725. If it was fixed  
> since then, I
> can easily recompile.
>
> The other party is on a recent revision from trunk and also runs  
> into the same
> problems in trying to call me via IPv6.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jason at jasonjgw.net  Mon Dec 15 15:29:56 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Dec 2008 10:29:56 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <0DEA56F4-9C15-47FA-8746-924CD6EF4109@freeswitch.org>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
	<20081215225451.GA5592@jdc.jasonjgw.net>
	<0DEA56F4-9C15-47FA-8746-924CD6EF4109@freeswitch.org>
Message-ID: <20081215232956.GA6139@jdc.jasonjgw.net>

On Mon, Dec 15, 2008 at 05:09:21PM -0600, Brian West wrote:
> sofia/internal-ipv6/888@[2001:470:7:ea::2]
> 
> (correct the profile for your needs)

That worked straight away.

At least it shows that there's nothing wrong with the IPv6 connectivity as
such. I'll try to isolate what is different between this and the call that I
was attempting yesterday.




From brian at freeswitch.org  Mon Dec 15 15:36:51 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Dec 2008 17:36:51 -0600
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081215232956.GA6139@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<081934B6-1899-4395-B907-5E80A4E00175@freeswitch.org>
	<20081215225451.GA5592@jdc.jasonjgw.net>
	<0DEA56F4-9C15-47FA-8746-924CD6EF4109@freeswitch.org>
	<20081215232956.GA6139@jdc.jasonjgw.net>
Message-ID: <2DC959F8-9A78-4D1D-8C6C-792A401689CE@freeswitch.org>

btw thats on an HE tunnel too ;)

/b

On Dec 15, 2008, at 5:29 PM, Jason White wrote:

> That worked straight away.
>
> At least it shows that there's nothing wrong with the IPv6  
> connectivity as
> such. I'll try to isolate what is different between this and the  
> call that I
> was attempting yesterday.




From c_cav_01 at yahoo.com  Mon Dec 15 21:21:35 2008
From: c_cav_01 at yahoo.com (ccav)
Date: Mon, 15 Dec 2008 21:21:35 -0800 (PST)
Subject: [Freeswitch-users]  Building a web config/billing gui
Message-ID: <21027515.post@talk.nabble.com>


For anyone interested, I'm in the process of building a web based
config/billing gui.  I could use some help though.  Anyone who has php
experience and knows the xml_curl interface pretty well and some spare time
to do some development would be a useful partner.

Also, if there's a resource online that defines all the params/variables and
subobjects for each of the fs object types, like directories, dialplans etc,
it would help me fill my parm database a lot faster.

For anyone interested in monitoring the development, it's at
www.sparkz.tv/smfs  login is demo:demo.  Feedback on features and
development direction is invited.  I want to get this done quickly.  I think
FS is awesome but no config/billing interface is going to stand in the way
of it's adoption so I'm burning the midnight oil to get this done.
-- 
View this message in context: http://www.nabble.com/Building-a-web-config-billing-gui-tp21027515p21027515.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From darren at aleph-com.net  Mon Dec 15 21:33:24 2008
From: darren at aleph-com.net (Darren Wiebe)
Date: Mon, 15 Dec 2008 22:33:24 -0700
Subject: [Freeswitch-users] Building a web config/billing gui
In-Reply-To: <21027515.post@talk.nabble.com>
References: <21027515.post@talk.nabble.com>
Message-ID: <49473DA4.7090501@aleph-com.net>

Are you interested in joining an existing project?  We've been working 
on the ASTPP freeswitch port for a number of months.   The billing is 
all in place but  help with the same stuff you're looking for would help 
to get things fleshed out a little more. 

Darren Wiebe
darren at aleph-com.net

ccav wrote:
> For anyone interested, I'm in the process of building a web based
> config/billing gui.  I could use some help though.  Anyone who has php
> experience and knows the xml_curl interface pretty well and some spare time
> to do some development would be a useful partner.
>
> Also, if there's a resource online that defines all the params/variables and
> subobjects for each of the fs object types, like directories, dialplans etc,
> it would help me fill my parm database a lot faster.
>
> For anyone interested in monitoring the development, it's at
> www.sparkz.tv/smfs  login is demo:demo.  Feedback on features and
> development direction is invited.  I want to get this done quickly.  I think
> FS is awesome but no config/billing interface is going to stand in the way
> of it's adoption so I'm burning the midnight oil to get this done.
>   




From msc at freeswitch.org  Mon Dec 15 21:37:01 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 15 Dec 2008 21:37:01 -0800
Subject: [Freeswitch-users] Building a web config/billing gui
In-Reply-To: <21027515.post@talk.nabble.com>
References: <21027515.post@talk.nabble.com>
Message-ID: <3C01F130-D55F-4A1A-888C-98C556363AF6@freeswitch.org>

Ccav,

Thanks for helping out with the project! If you haven't already joined  
us on the irc channel please do so: #freeswitch on irc.freenode.net.  
Another channel you might be interested in is #tcapi. There is a group  
working on a general purpose GUI for FreeSWITCH at tcapi.org. They've  
probably faced some of the same challenges that you have.

We will be glad to help you however we possibly can.

-MC

Sent from my iPhone

On Dec 15, 2008, at 9:21 PM, ccav <c_cav_01 at yahoo.com> wrote:

>
> For anyone interested, I'm in the process of building a web based
> config/billing gui.  I could use some help though.  Anyone who has php
> experience and knows the xml_curl interface pretty well and some  
> spare time
> to do some development would be a useful partner.
>
> Also, if there's a resource online that defines all the params/ 
> variables and
> subobjects for each of the fs object types, like directories,  
> dialplans etc,
> it would help me fill my parm database a lot faster.
>
> For anyone interested in monitoring the development, it's at
> www.sparkz.tv/smfs  login is demo:demo.  Feedback on features and
> development direction is invited.  I want to get this done quickly.   
> I think
> FS is awesome but no config/billing interface is going to stand in  
> the way
> of it's adoption so I'm burning the midnight oil to get this done.
> -- 
> View this message in context: http://www.nabble.com/Building-a-web-config-billing-gui-tp21027515p21027515.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Prometheus001 at gmx.net  Mon Dec 15 23:59:26 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 16 Dec 2008 08:59:26 +0100
Subject: [Freeswitch-users] mod_shout and mp3 formats
Message-ID: <49475FDE.7080108@gmx.net>

I try to play mp3 I generated through Cepstral TTs and which I encoded
via lame.
However they won't play, so my question is: Which mp3 formats are supported?

I generate the wav files by the following
/opt/swift/bin/swift -n Katrin -p
audio/channels=1,cst/f0_shift=.8,speech/rate=120,audio/sampling-rate=8000,audio/deadair=2
-o $wavefile $text

Then I convert to mp3 by the following variations:
lame 46.wav 46.mp3
lame -s 32 46.wav 46.mp3
lame --preset 128 46.wav 46.mp3
lame --resample 44.1 --preset 128 46.wav 46.mp3
lame --resample 32 --preset 128 46.wav 46.mp3
lame --resample 44.1 46.wav 46.mp3
lame --resample 44.1 -m s --preset 128 46.wav 46.mp3
lame --resample 44.1 -m s  46.wav 46.mp3
lame --resample 44.1 -m s -b 128 46.wav 46.mp3
lame --resample 44.1 -m s -B 24 46.wav 46.mp3
lame --preset voice -v -B 64 -a 46.wav 46.mp3

None of them worked with the playback application
(shout://localhost/tts/46.mp3). The sound files had a length of between
2 and 5 sec. 2 Times during various tries they played at least
partially. But at the next try they didn't play again. However I have a
prerecorded sound file (44.1KHz, 128 kBits stereo music) which always
plays well.
The console shows me that all files are successfully played and I get a
channel_ececute and a channel_ececute_complete after some seconds during
event_socket. But I don't hear any sound.

All above samples however played well with Totem on Ubuntu.

The wiki tells me that almost any mp3 format should play. What am I
doing wrong here?

Another question: Should normal wav files play as well? Also with wav I
cannot hear any sound.

Best regards
Peter







From Prometheus001 at gmx.net  Tue Dec 16 00:06:16 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 16 Dec 2008 09:06:16 +0100
Subject: [Freeswitch-users] running custom script with bind_meta_app
In-Reply-To: <11990ade0812150925i33e9cf6ex6a1fe53c423fae9b@mail.gmail.com>
References: <11990ade0812142332h2efbe826ob6e4dded694a6baf@mail.gmail.com>	<910AFB7A-9CDE-49AE-A72D-7942F8F79DA3@freeswitch.org>
	<11990ade0812150925i33e9cf6ex6a1fe53c423fae9b@mail.gmail.com>
Message-ID: <49476178.7080603@gmx.net>

I use Telegraph with Ruby on Rails to listen on the event socket
interface. With Telegraph you can register on any FS event and get all
channel variables in a hash for further processing. Interactions then
can be done via event_socket intreface.
Telegraph is not finished yet, but for me it was a good and easy point
to start.

Best regards
Peter

Stephen Crosby schrieb:
> I just want to listen for some DTMF sequence while in a conference.
> The conference host should be able to enter the sequence at any time
> (and any number of times) to run a custom script. I've already written
> one in javascript, but I can rewrite it in another language if it's
> easier.
>
> On Mon, Dec 15, 2008 at 7:09 AM, Brian West <brian at freeswitch.org> wrote:
>   
>> What are you wanting to accomplish first?
>> /b
>> On Dec 15, 2008, at 1:32 AM, Stephen Crosby wrote:
>>
>> I'm Stephen Crosby, and I've just started working with freeswitch.
>> It's been great so far.
>>
>> I want to run a custom script inside a conference when a DTMF sequence
>> is entered. I found bind_meta_app and thought it would be perfect, but
>> I can't seem to get it to work. When I dial-in and press *8, I get no
>> debugging output at all. When I press another sequence like *9 for
>> instance, I get: [WARNING] switch_ivr_async.c:1429 meta_on_dtmf()
>> sofia/external/5593495805 at sip.gafachi.com Ignoring meta digit '9' not
>> mapped. The script I wrote has been tested with "jsrun script.js" from
>> the command line and it does work. I've got the debugging level all
>> the way up and there's just not much for me to go on. Any help would
>> be greatly appreciated.
>>
>> <extension name="gafachi-DID1">
>>  <condition field="destination_number" expression="^xxxxxxxxxxx$">
>>    <action application="answer"/>
>>    <action application="bind_meta_app" data="8 ab s
>> javascript::$${base_dir}/scripts/script.js"/>
>>    <action application="conference" data="3000 at default"/>
>>  </condition>
>> </extension>
>>
>> --Stephen
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From jonas.gauffin at gmail.com  Tue Dec 16 00:07:30 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 16 Dec 2008 09:07:30 +0100
Subject: [Freeswitch-users] Bridging through gateway
In-Reply-To: <49468B9D.5010903@freeswitch.org>
References: <d021d3be0812150219k655e201l190479e048a7d2e3@mail.gmail.com>
	<20081215110346.GA12681@jdc.jasonjgw.net>
	<432F2238-BF29-4E6F-8B7A-BDB29C49B27A@freeswitch.org>
	<d021d3be0812150801g3d904fb5xe98b281f142a2d95@mail.gmail.com>
	<49468B9D.5010903@freeswitch.org>
Message-ID: <d021d3be0812160007s42a6979bn22f1601c59d862fa@mail.gmail.com>

        <?xml version="1.0" encoding="UTF-8" standalone="no"?>
                <document type="freeswitch/xml">
                  <section name="dialplan" description="ModCurlDialplan">
                    <context name="default">
                      <extension name=" 0236661201 ">
                        <condition field="destination_number"
expression="^0236661201$">
                          <action application="set" data="gate_site_id=1"/>
                          <action application="set"
data="effective_caller_id_number=0812024481"/>
                          <action application="set"
data="gate_external_destination=+46236661201"/>
                          <action application="bridge" data="sofia/default/
0236661201 at sip-corporate2.tele2.se" />
                          <action application="hangup"
data="NORMAL_CLEARING"/>
                        </condition>
                      </extension>
                    </context>
                  </section>
                </document>


That's it. the sofia configs are pretty much default.
On Mon, Dec 15, 2008 at 5:53 PM, Raymond Chandler <
intralanman at freeswitch.org> wrote:

>  posting relevant pieces of your dialplan and sofia configs would probably
> help a bit.
>
> -Ray
>
>
> Jonas Gauffin wrote:
>
> Yeah I know. That's what I'm trying to do, but with the domain name (
> sip-corporate2.tele2.se) instead of the  ip.
>  I'm not sure that the gateway works without authentication and I'm
> wondering if MANDATORY_IE_MISSING means that the gateway wants
> authentication or if it means something else.
> It's the "No Matching gateway found" message that is confusing, since I'm
> not trying to use a registered gateway?
>
> On Mon, Dec 15, 2008 at 4:05 PM, Brian West <brian at freeswitch.org> wrote:
>
>> If you don't need auth you don't need a gateway.  sofia/profile/
>> number at remoteip is all you should need.
>>
>> /b
>>
>> On Dec 15, 2008, at 5:03 AM, Jason White wrote:
>>
>> > On Mon, Dec 15, 2008 at 11:19:44AM +0100, Jonas Gauffin wrote:
>> >> I'm trying to bridge using a non-registered gateway. And I
>> >> get MANDATORY_IE_MISSING back. Why is that?
>> >
>> > Does the gateway allow unauthenticated clients to make calls? If you
>> > obtain a
>> > SIP trace, you'll be able to see whether it's an authentication issue.
>> >
>> > As an aside, it would be an improvement to FreeSWITCH if Sofia
>> > debugging could
>> > be turned on and off within a running FreeSWITCH instance, including
>> > SIP
>> > traces, instead of the administrator's having to restart FreeSWITCH
>> > with
>> > environment variables exported, as is presently required according
>> > to the
>> > wiki.
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fidibus83 at aol.com  Tue Dec 16 01:54:47 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 16 Dec 2008 10:54:47 +0100
Subject: [Freeswitch-users] Zaptel Error!!!
Message-ID: <002101c95f64$56958e60$6445310a@Franzi>

Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

*

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg ?vv:

 

31 channels configured.

 

ioctl(ZT_LOADZONE) failed: Invalid argument

Notice: Configuration file is /etc/zaptel.conf

line 288: Unable to register tone zone 'de'

 

 

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

What do I wrong? Please help me!
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From mszlazak at aol.com  Tue Dec 16 02:02:25 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Tue, 16 Dec 2008 05:02:25 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
Message-ID: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>

I'm making a call internally from a soft phone to an extension that is suppose to bridge the call internally to another application on the same computer. The applications logs indicate that a connection was made but sound is not being passed back from the application through freeswitch to the softphone. There maybe an issue with rtp timing and associated ports but I'm very new at diagnosing this and fixing the problem.

I've attached both a copy of the FS log and an associated pcap file.

It's all on Windows XP. 

Could someone please take a look.

Thanks.
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From jonas.gauffin at gmail.com  Tue Dec 16 02:54:40 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 16 Dec 2008 11:54:40 +0100
Subject: [Freeswitch-users] Bind error
Message-ID: <d021d3be0812160254y1e0ef82cmc83ae914a0b5324a@mail.gmail.com>

I get a bind error for the RTP, can someone be kind and explain why?

2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2388 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000]/[PCMA:8:8000]
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1596 sofia_glue_tech_set_codec()
Set Codec sofia/internal/0236661201 at sip-corporate2.tele2.se PCMA/8000 20 ms
160 samples
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2352 sofia_glue_negotiate_sdp() Set
2833 dtmf payload to 101
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1814 sofia_glue_activate_rtp()
AUDIO RTP [sofia/internal/0236661201 at sip-corporate2.tele2.se] 192.168.1.112
port 17102 -> 130.244.5X.XX port 27354 codec: 8 ms: 20
2008-12-16 10:47:07 [DEBUG] switch_rtp.c:858 switch_rtp_create() Starting
timer [soft] 160 bytes per 20000ms
2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Ring-Ready sofia/internal/0236661201 at sip-corporate2.tele2.se!
2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Pre-Answer sofia/internal/0236661201 at sip-corporate2.tele2.se!
2008-12-16 10:47:07 [DEBUG] switch_channel.c:1615
switch_channel_perform_pre_answer()
sofia/external/0707992871 at 212.151.14X.XXX receive message
[SWITCH_MESSAGE_INDICATE_PROGRESS]
2008-12-16 10:47:07 [INFO] mod_sofia.c:1253 sofia_receive_message() Asked to
send early media by sofia/external/0707992871 at 212.151.14X.XXX
2008-12-16 10:47:07 [DEBUG] switch_channel.c:1585
switch_channel_perform_mark_pre_answered() Send signal
sofia/external/0707992871 at 212.151.14X.XXX [BREAK]
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup()
STUN Success [85.89.7X.XXX]:[33594]
2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1814 sofia_glue_activate_rtp()
AUDIO RTP [sofia/external/0707992871 at 212.151.14X.XXX] 192.168.1.112 port
17108 -> 130.244.1X.XXX port 17148 codec: 8 ms: 20
2008-12-16 10:47:07 [DEBUG] switch_rtp.c:858 switch_rtp_create() Starting
timer [soft] 160 bytes per 20000ms

2008-12-16 10:47:07 [ERR] sofia_glue.c:2045 sofia_glue_activate_rtp() AUDIO
RTP REPORTS ERROR: [Bind Error!]

2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2046 sofia_glue_activate_rtp()
Hangup sofia/external/0707992871 at 212.151.14X.XXX [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
2008-12-16 10:47:07 [DEBUG] switch_channel.c:1478
switch_channel_perform_hangup() Send signal
sofia/external/0707992871 at 212.151.14X.XXX [KILL]
2008-12-16 10:47:07 [DEBUG] switch_core_session.c:806
switch_core_session_signal_state_change() Send signal
sofia/external/0707992871 at 212.151.14X.XXX [BREAK]
2008-12-16 10:47:07 [INFO] mod_sofia.c:1294 sofia_receive_message() Ring
SDP:
v=0
o=FreeSWITCH 1229387233 1229387234 IN IP4 85.89.7X.XXX
s=FreeSWITCH
c=IN IP4 85.89.7X.XXX
t=0 0
m=audio 33594 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2008-12-16 10:47:07 [NOTICE] mod_sofia.c:1297 sofia_receive_message()
Ring-Ready sofia/external/0707992871 at 212.151.14X.XXX!
2008-12-16 10:47:07 [NOTICE] mod_sofia.c:1297 sofia_receive_message()
Pre-Answer sofia/external/0707992871 at 212.151.14X.XXX!
2008-12-16 10:47:07 [DEBUG] switch_core_session.c:510
switch_core_session_perform_receive_message() Send signal
sofia/external/0707992871 at 212.151.14X.XXX [BREAK]
2008-12-16 10:47:07 [DEBUG] switch_ivr_originate.c:1625
switch_ivr_originate() Originate Resulted in Error Cause: 487
[ORIGINATOR_CANCEL]
2008-12-16 10:47:07 [NOTICE] switch_ivr_originate.c:1666
switch_ivr_originate() Hangup sofia/internal/
0236661201 at sip-corporate2.tele2.se [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2008-12-16 10:47:07 [DEBUG] switch_channel.c:1478
switch_channel_perform_hangup() Send signal sofia/internal/
0236661201 at sip-corporate2.tele2.se [KILL]
2008-12-16 10:47:07 [DEBUG] switch_core_session.c:806
switch_core_session_signal_state_change() Send signal sofia/internal/
0236661201 at sip-corporate2.tele2.se [BREAK]
2008-12-16 10:47:07 [INFO] mod_dptools.c:1869 audio_bridge_function()
Originate Failed.  Cause: ORIGINATOR_CANCEL
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From scott.ellis at novatex.com.au  Tue Dec 16 03:03:43 2008
From: scott.ellis at novatex.com.au (Scott Ellis)
Date: Tue, 16 Dec 2008 22:03:43 +1100
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
Message-ID: <49478B0F.3000802@novatex.com.au>

I have a standard install, and I am trying to get a Pennytel gateway to 
register.

After looking at Wireshark traces of x-lite registering and FreeSwitch 
registering, FreeSwitch is not sending any authentication information 
with the registration request. I am obviously missing something here!

I understand for incoming calls you don't want authentication, but for 
outgoing it is obviously required.

Is there a flag somewhere that I am supposed to set? The file was taken 
from the wiki page, and looks like it was previously tested when using 
the obsolete outbound directory structure.

The following file is in the conf/sip_profiles/external directory.

<include>
  <gateway name="PennyTel">
    <param name="username" value="888917xxxx"/>
    <param name="password" value="xxxxxxx"/>
    <param name="realm" value="sip.pennytel.com"/>
    <param name="proxy" value="sip.pennytel.com"/>
    <param name="register" value="true"/>
    <param name="expire-seconds" value="60"/>
  </gateway>
</include>

Thanks.

Scott




From fidibus83 at aol.com  Tue Dec 16 03:34:49 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 16 Dec 2008 12:34:49 +0100
Subject: [Freeswitch-users]  Openzap ERROR can't dial
Message-ID: <004a01c95f72$4e59aca0$6445310a@Franzi>

Have nobody an idea?

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From helmut.kuper at ewetel.de  Tue Dec 16 04:32:23 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 13:32:23 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <493F7148.40705@ewetel.de>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>
	<493F7148.40705@ewetel.de>
Message-ID: <49479FD7.2070200@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

hm have libtiff4 installed. aptitude says  this:

v   libtiff-dev                                                     -
p   libtiff-opengl                                                  -
TIFF manipulation and conversion tools
p   libtiff-tools                                                   -
TIFF manipulation and conversion tools
i A libtiff4                                                        -
Tag Image File Format (TIFF) library
i A libtiff4-dev                                                    -
Tag Image File Format library (TIFF), development files
i A libtiffxx0c2                                                    -
Tag Image File Format (TIFF) library -- C++ interface

so it is there. but still I get this error during compiling:

Compiling mod_fax.c...
In file included from /usr/include/spandsp.h:97,
                 from mod_fax.c:36:
../../../../libs/spandsp/src/spandsp/t38_gateway.h:94: error: expected
specifier-qualifier-list before ?fax_modems_state_t?
../../../../libs/spandsp/src/spandsp/t38_gateway.h:204: error: expected
specifier-qualifier-list before ?t38_non_ecm_buffer_state_t?
In file included from /usr/include/spandsp.h:99,
                 from mod_fax.c:36:
../../../../libs/spandsp/src/spandsp/t31.h:57: error: expected
specifier-qualifier-list before ?fax_modems_state_t?
In file included from /usr/include/spandsp.h:100,
                 from mod_fax.c:36:
../../../../libs/spandsp/src/spandsp/fax.h:52: error: expected
specifier-qualifier-list before ?fax_modems_state_t?

I still have to do a configure and a make in libs/spandsp befor
compiling FS.


regards
helmut
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From vkobashi at ydeasolutions.com.br  Tue Dec 16 04:59:11 2008
From: vkobashi at ydeasolutions.com.br (vinicius)
Date: Tue, 16 Dec 2008 10:59:11 -0200
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <494198F3.10806@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>
	<494198F3.10806@ydeasolutions.com.br>
Message-ID: <4947A61F.6060806@ydeasolutions.com.br>

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From helmut.kuper at ewetel.de  Tue Dec 16 05:06:59 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 14:06:59 +0100
Subject: [Freeswitch-users] mod_ldap
Message-ID: <4947A7F3.90703@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I updated to latest FS trunk today and tried to compile mod_ldap and got
this errors:


mkdir .libs
ar cru .libs/libldap.a  bind.o open.o result.o error.o compare.o
search.o controls.o messages.o references.o extended.o cyrus.o modify.o
add.o modrdn.o delete.o abandon.o sasl.o sbind.o kbind.o unbind.o
cancel.o filter.o free.o sort.o passwd.o whoami.o getdn.o getentry.o
getattr.o getvalues.o addentry.o request.o os-ip.o url.o sortctrl.o
vlvctrl.o init.o options.o print.o string.o util-int.o schema.o
charray.o tls.o os-local.o dnssrv.o utf-8.o utf-8-conv.o turn.o
groupings.o txn.o ppolicy.o version.o
ranlib .libs/libldap.a
creating libldap.la
(cd .libs && rm -f libldap.la && ln -s ../libldap.la libldap.la)
cc -g -O2 -o apitest apitest.o  ./.libs/libldap.a
/opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/liblber/.libs/liblber.a
../../libraries/liblber/.libs/liblber.a
../../libraries/liblutil/liblutil.a /usr/lib/libsasl2.a -ldl -lresolv
./.libs/libldap.a(os-ip.o): In function `ldap_pvt_is_socket_ready':
/opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194:
warning: `sys_errlist' is deprecated; use `strerror' or `strerror_r' instead
/opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194:
warning: `sys_nerr' is deprecated; use `strerror' or `strerror_r' instead
/usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_close':
(.text+0x348): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
(.text+0x3d0): undefined reference to `db_create'
/usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
(.text+0x453): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_getdata':
(.text+0x7e8): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
(.text+0xa4e): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
(.text+0xafa): undefined reference to `db_strerror'
/usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
(.text+0x6c8): undefined reference to `DES_key_sched'
/usr/lib/libsasl2.a(digestmd5.o): In function `init_des':


When I change src/mod/directories/mod_ldap/Makefile line 3 to
"LDAP=openldap-2.4.11" and putting openldap-2.4.11.tgz in libs/ manually
it compiles without error, but with some warnings.


Maybe the openldap-2.3.19 delivered by FS is too old?

regards
helmut
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From carole.olivier at enst.fr  Tue Dec 16 05:07:47 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Tue, 16 Dec 2008 05:07:47 -0800 (PST)
Subject: [Freeswitch-users]  general question about API command
Message-ID: <21032754.post@talk.nabble.com>


Hello,

I have a general question about the API commands. Some of them are not
available in the dialplan like uuid_transfer. I was wondering how to call an
API command without using the CLI. Especially I would be interested in
knowing if there is any way to call them from a phone, I mean bind a key to
an API command.

For instance I would like to transfer both members of a simple call into a
conference by dialing *1. 
I have seen the transfer is possible from the CLI by doing:
api uuid_transfer <uuid> -both 3001
but I do not know how to do it else.

If somebody could give me an insight about the topic it would be great.
Thanks,
Carole








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View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
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From mike at jerris.com  Tue Dec 16 05:19:51 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:19:51 -0500
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <002101c95f64$56958e60$6445310a@Franzi>
References: <002101c95f64$56958e60$6445310a@Franzi>
Message-ID: <7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com>

It sounds like there is no de tonezone in the zaptel drivers, but I  
can't imagine thats true.  What version of the drivers do you have  
installed?


On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:

> Hello, I?m a newbie in FS and my English isn?t very good but I try  
> to explain my problem. Hopefully you can understand me. J
>
> I have a Linux-Server with a Digium Wildcard TE110P. I install and  
> configure openzap (PRI/ISDN) and zaptel. But I have an error when I  
> execute ztcfg ?vv:
>
> 31 channels configured.
>
> ioctl(ZT_LOADZONE) failed: Invalid argument
> Notice: Configuration file is /etc/zaptel.conf
> line 288: Unable to register tone zone 'de'
>
>
>
> ZAPTEL.CONF
>
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
> loadzone = de
> defaultzone=de
>
> What do I wrong? Please help me!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Dec 16 05:24:16 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:24:16 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
In-Reply-To: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>
References: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>
Message-ID: <957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com>

If its all local you can also just use:

http://wiki.freeswitch.org/wiki/Bypass_Media

If your still trying to figure it out it could be any number of  
things, but most relating to misconfigured endpoints or freeswitch,  
take a look at the sip trace and make sure everything is using the  
right ip addresses instead of using internal when they should be  
external or the other way around.

Mike

On Dec 16, 2008, at 5:02 AM, mszlazak at aol.com wrote:

> I'm making a call internally from a soft phone to an extension that  
> is suppose to bridge the call internally to another application on  
> the same computer. The applications logs indicate that a connection  
> was made but sound is not being passed back from the application  
> through freeswitch to the softphone. There maybe an issue with rtp  
> timing and associated ports but I'm very new at diagnosing this and  
> fixing the problem.
>
> I've attached both a copy of the FS log and an associated pcap file.
>
> It's all on Windows XP.
>
> Could someone please take a look.
>
> Thanks.
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
> <free.zip>_______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Dec 16 05:25:12 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:25:12 -0500
Subject: [Freeswitch-users] Bind error
In-Reply-To: <d021d3be0812160254y1e0ef82cmc83ae914a0b5324a@mail.gmail.com>
References: <d021d3be0812160254y1e0ef82cmc83ae914a0b5324a@mail.gmail.com>
Message-ID: <85B9BF33-5F2D-4674-9840-069D7D1BB99C@jerris.com>


On Dec 16, 2008, at 5:54 AM, Jonas Gauffin wrote:

> I get a bind error for the RTP, can someone be kind and explain why?
>

Some other program is using the ports already?



From mike at jerris.com  Tue Dec 16 05:26:30 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:26:30 -0500
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
In-Reply-To: <49478B0F.3000802@novatex.com.au>
References: <49478B0F.3000802@novatex.com.au>
Message-ID: <153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>

We send authentication after we get a challenge because on startup we  
need the nonce from them to build the hash in the Auth header properly.

Mike

On Dec 16, 2008, at 6:03 AM, Scott Ellis wrote:

> I have a standard install, and I am trying to get a Pennytel gateway  
> to
> register.
>
> After looking at Wireshark traces of x-lite registering and FreeSwitch
> registering, FreeSwitch is not sending any authentication information
> with the registration request. I am obviously missing something here!
>
> I understand for incoming calls you don't want authentication, but for
> outgoing it is obviously required.
>
> Is there a flag somewhere that I am supposed to set? The file was  
> taken
> from the wiki page, and looks like it was previously tested when using
> the obsolete outbound directory structure.
>
> The following file is in the conf/sip_profiles/external directory.
>
> <include>
>  <gateway name="PennyTel">
>    <param name="username" value="888917xxxx"/>
>    <param name="password" value="xxxxxxx"/>
>    <param name="realm" value="sip.pennytel.com"/>
>    <param name="proxy" value="sip.pennytel.com"/>
>    <param name="register" value="true"/>
>    <param name="expire-seconds" value="60"/>
>  </gateway>
> </include>
>
> Thanks.
>
> Scott
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Tue Dec 16 05:27:10 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:27:10 -0500
Subject: [Freeswitch-users] Openzap ERROR can't dial
In-Reply-To: <004a01c95f72$4e59aca0$6445310a@Franzi>
References: <004a01c95f72$4e59aca0$6445310a@Franzi>
Message-ID: <488DE1CC-8F76-49AE-90EF-ED27263AB134@jerris.com>


On Dec 16, 2008, at 6:34 AM, fidibus83 wrote:

> Have nobody an idea?
>

Idea about what?
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From mike at jerris.com  Tue Dec 16 05:29:19 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:29:19 -0500
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <49479FD7.2070200@ewetel.de>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>
	<493F7148.40705@ewetel.de> <49479FD7.2070200@ewetel.de>
Message-ID: <0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>

The only reason that I can come up with would be if configure is not  
detecting it for whatever reason.  Check the config.log in the main fs  
src dir to see if it mentions the check at all.

MIke

On Dec 16, 2008, at 7:32 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> hm have libtiff4 installed. aptitude says  this:
>
> v   libtiff-dev                                                     -
> p   libtiff-opengl                                                  -
> TIFF manipulation and conversion tools
> p   libtiff-tools                                                   -
> TIFF manipulation and conversion tools
> i A libtiff4                                                        -
> Tag Image File Format (TIFF) library
> i A libtiff4-dev                                                    -
> Tag Image File Format library (TIFF), development files
> i A libtiffxx0c2                                                    -
> Tag Image File Format (TIFF) library -- C++ interface
>
> so it is there. but still I get this error during compiling:
>
> Compiling mod_fax.c...
> In file included from /usr/include/spandsp.h:97,
>                 from mod_fax.c:36:
> ../../../../libs/spandsp/src/spandsp/t38_gateway.h:94: error: expected
> specifier-qualifier-list before ?fax_modems_state_t?
> ../../../../libs/spandsp/src/spandsp/t38_gateway.h:204: error:  
> expected
> specifier-qualifier-list before ?t38_non_ecm_buffer_state_t?
> In file included from /usr/include/spandsp.h:99,
>                 from mod_fax.c:36:
> ../../../../libs/spandsp/src/spandsp/t31.h:57: error: expected
> specifier-qualifier-list before ?fax_modems_state_t?
> In file included from /usr/include/spandsp.h:100,
>                 from mod_fax.c:36:
> ../../../../libs/spandsp/src/spandsp/fax.h:52: error: expected
> specifier-qualifier-list before ?fax_modems_state_t?
>
> I still have to do a configure and a make in libs/spandsp befor
> compiling FS.




From mike at jerris.com  Tue Dec 16 05:32:02 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 08:32:02 -0500
Subject: [Freeswitch-users] mod_ldap
In-Reply-To: <4947A7F3.90703@ewetel.de>
References: <4947A7F3.90703@ewetel.de>
Message-ID: <818373FA-A5DE-4534-B807-CDE119E9EE42@jerris.com>

Can you please file a bug in jira on this issue, we can get this  
corrected today.
On Dec 16, 2008, at 8:06 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I updated to latest FS trunk today and tried to compile mod_ldap and  
> got
> this errors:
>
>
> mkdir .libs
> ar cru .libs/libldap.a  bind.o open.o result.o error.o compare.o
> search.o controls.o messages.o references.o extended.o cyrus.o  
> modify.o
> add.o modrdn.o delete.o abandon.o sasl.o sbind.o kbind.o unbind.o
> cancel.o filter.o free.o sort.o passwd.o whoami.o getdn.o getentry.o
> getattr.o getvalues.o addentry.o request.o os-ip.o url.o sortctrl.o
> vlvctrl.o init.o options.o print.o string.o util-int.o schema.o
> charray.o tls.o os-local.o dnssrv.o utf-8.o utf-8-conv.o turn.o
> groupings.o txn.o ppolicy.o version.o
> ranlib .libs/libldap.a
> creating libldap.la
> (cd .libs && rm -f libldap.la && ln -s ../libldap.la libldap.la)
> cc -g -O2 -o apitest apitest.o  ./.libs/libldap.a
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/ 
> liblber/.libs/liblber.a
> ../../libraries/liblber/.libs/liblber.a
> ../../libraries/liblutil/liblutil.a /usr/lib/libsasl2.a -ldl -lresolv
> ./.libs/libldap.a(os-ip.o): In function `ldap_pvt_is_socket_ready':
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/ 
> libldap/os-ip.c:194:
> warning: `sys_errlist' is deprecated; use `strerror' or `strerror_r'  
> instead
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/ 
> libldap/os-ip.c:194:
> warning: `sys_nerr' is deprecated; use `strerror' or `strerror_r'  
> instead
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_close':
> (.text+0x348): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
> (.text+0x3d0): undefined reference to `db_create'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
> (.text+0x453): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_getdata':
> (.text+0x7e8): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
> (.text+0xa4e): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
> (.text+0xafa): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
> (.text+0x6c8): undefined reference to `DES_key_sched'
> /usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
>
>
> When I change src/mod/directories/mod_ldap/Makefile line 3 to
> "LDAP=openldap-2.4.11" and putting openldap-2.4.11.tgz in libs/  
> manually
> it compiles without error, but with some warnings.
>
>
> Maybe the openldap-2.3.19 delivered by FS is too old?
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklHp/MACgkQ4tZeNddg3dzjOgCfSExGU7PSWSdAxI4VCZIosrLJ
> 25kAoK0igc53iV1PhWkj5faAVOL78E8B
> =ny1E
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Tue Dec 16 05:48:55 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Dec 2008 07:48:55 -0600
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <21032754.post@talk.nabble.com>
References: <21032754.post@talk.nabble.com>
Message-ID: <191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>

There is a transfer dial plan application also.

There is not really any benefit in blocking the api commands from the
dialplan
apart from the potential for a blocking api call to delay the audio stream
which
you can do at your own risk and use the sleep application with 0ms to flush
the read buffers.

So I guess i can lift that limitation in tree.

see r10790


On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> Hello,
>
> I have a general question about the API commands. Some of them are not
> available in the dialplan like uuid_transfer. I was wondering how to call
> an
> API command without using the CLI. Especially I would be interested in
> knowing if there is any way to call them from a phone, I mean bind a key to
> an API command.
>
> For instance I would like to transfer both members of a simple call into a
> conference by dialing *1.
> I have seen the transfer is possible from the CLI by doing:
> api uuid_transfer <uuid> -both 3001
> but I do not know how to do it else.
>
> If somebody could give me an insight about the topic it would be great.
> Thanks,
> Carole
>
>
>
>
>
>
>
>
> --
> View this message in context:
> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From fidibus83 at aol.com  Tue Dec 16 05:54:01 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 16 Dec 2008 14:54:01 +0100
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com>
References: <002101c95f64$56958e60$6445310a@Franzi>
	<7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com>
Message-ID: <007601c95f85$c1462aa0$6445310a@Franzi>

I have installed zaptel-1.4.11 

I have looked in zonedata.c and there is configured de-tonezone 

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael Jerris
Gesendet: Dienstag, 16. Dezember 2008 14:20
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

It sounds like there is no de tonezone in the zaptel drivers, but I can't
imagine thats true.  What version of the drivers do you have installed?  

 

 

On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:





Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

 

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg ?vv:

 

31 channels configured.

 

ioctl(ZT_LOADZONE) failed: Invalid argument

Notice: Configuration file is /etc/zaptel.conf

line 288: Unable to register tone zone 'de'

 

 

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

What do I wrong? Please help me!

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

=
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From helmut.kuper at ewetel.de  Tue Dec 16 06:35:08 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 15:35:08 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>
	<49479FD7.2070200@ewetel.de>
	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>
Message-ID: <4947BC9C.8040302@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

hm no, there is nothing like libtiff, spandsp or even fax to find in
config.log.

regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.9 (MingW32)

iEYEARECAAYFAklHvJsACgkQ4tZeNddg3dwsOwCeI1hVTC3CeW/SLiuHn0g6VTYW
1WgAn1dOjzoBgU8Ln6Wri/a53O8rRZOO
=fHh+
-----END PGP SIGNATURE-----



From msc at freeswitch.org  Tue Dec 16 06:37:26 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Dec 2008 06:37:26 -0800
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <007601c95f85$c1462aa0$6445310a@Franzi>
References: <002101c95f64$56958e60$6445310a@Franzi>
	<7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com>
	<007601c95f85$c1462aa0$6445310a@Franzi>
Message-ID: <9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org>

Just a hunch but try removing the spaces in this line:

loadzone=de

Zaptel can be quirky.
-MC

Sent from my iPhone

On Dec 16, 2008, at 5:54 AM, "fidibus83" <fidibus83 at aol.com> wrote:

> I have installed zaptel-1.4.11
>
> I have looked in zonedata.c and there is configured de-tonezone
>
>
>
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] Im Auftrag von Michael Jerris
> Gesendet: Dienstag, 16. Dezember 2008 14:20
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Zaptel Error!!!
>
>
>
> It sounds like there is no de tonezone in the zaptel drivers, but I  
> can't imagine thats true.  What version of the drivers do you have  
> installed?
>
>
>
>
>
> On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:
>
>
>
>
> Hello, I?m a newbie in FS and my English isn?t very good but I  
> try to explain my problem. Hopefully you can understand me. J
>
>
>
> I have a Linux-Server with a Digium Wildcard TE110P. I install and  
> configure openzap (PRI/ISDN) and zaptel. But I have an error when I  
> execute ztcfg ?vv:
>
>
>
> 31 channels configured.
>
>
>
> ioctl(ZT_LOADZONE) failed: Invalid argument
>
> Notice: Configuration file is /etc/zaptel.conf
>
> line 288: Unable to register tone zone 'de'
>
>
>
>
>
>
>
> ZAPTEL.CONF
>
>
>
> span=1,1,0,ccs,hdb3,crc4
>
> bchan=1-15,17-31
>
> dchan=16
>
> loadzone = de
>
> defaultzone=de
>
>
>
> What do I wrong? Please help me!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> =
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From carole.olivier at enst.fr  Tue Dec 16 06:43:32 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Tue, 16 Dec 2008 06:43:32 -0800 (PST)
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
Message-ID: <21033993.post@talk.nabble.com>


Thanks for this answer. 
Just a question so that everything is clear: there is no command to bridge
directly a call and both legs into a conference? We have to transfer the
call to another extension and from there create the conference isn't?

I used the command transfer but I still have a problem. I do the following
and it did not work: in my dialplan I write:
<action application="bind_meta_app" data="1 ab s transfer::3333"/>

The extension 3333 works fine.

However, nothing happens when I press *1.
Do you have an idea where am I wrong?

(A subsidiary question: will both legs be transferred to the extension 3333
or just the one which press *1? is there a way to transfer both together?)


Thanks a lot,
Carole


Anthony Minessale-2 wrote:
> 
> There is a transfer dial plan application also.
> 
> There is not really any benefit in blocking the api commands from the
> dialplan
> apart from the potential for a blocking api call to delay the audio stream
> which
> you can do at your own risk and use the sleep application with 0ms to
> flush
> the read buffers.
> 
> So I guess i can lift that limitation in tree.
> 
> see r10790
> 
> 
> On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr> wrote:
> 
>>
>> Hello,
>>
>> I have a general question about the API commands. Some of them are not
>> available in the dialplan like uuid_transfer. I was wondering how to call
>> an
>> API command without using the CLI. Especially I would be interested in
>> knowing if there is any way to call them from a phone, I mean bind a key
>> to
>> an API command.
>>
>> For instance I would like to transfer both members of a simple call into
>> a
>> conference by dialing *1.
>> I have seen the transfer is possible from the CLI by doing:
>> api uuid_transfer <uuid> -both 3001
>> but I do not know how to do it else.
>>
>> If somebody could give me an insight about the topic it would be great.
>> Thanks,
>> Carole
>>
>>
>>
>>
>>
>>
>>
>>
>> --
>> View this message in context:
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From helmut.kuper at ewetel.de  Tue Dec 16 06:45:34 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 15:45:34 +0100
Subject: [Freeswitch-users] OpenZAP:  "Received unhandled message 125"
In-Reply-To: <EBD35DE7-A6C9-4072-AC8E-F2D49E7098C8@freeswitch.org>
References: <492ED886.9020805@gmx.net>
	<EBD35DE7-A6C9-4072-AC8E-F2D49E7098C8@freeswitch.org>
Message-ID: <4947BF0E.50702@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I get that "CRIT" error on my q931 pri as well. I found that the message
is a Q931 status message from remote end and I guess it only will tell
you the call state of your outgoing call at the remote end to make sure
you are in same state. So it is a nice to have to get this mesage
decoded for debugging reasons.

regards
helmut



Am 27.11.2008 19:02, schrieb Michael S Collins:
> You can ignore this one for now. Eventually this will be handled but  
> it shouldn't affect your calls. I've been ignoring it for six months. :)
> 
> -MC
> 
> Sent from my iPhone
> 
> On Nov 27, 2008, at 9:27 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> 
>> I have installed OpenZAP with a TE220 card and EuroISDN.
>>
>> When I bridge an outgoing call I get a "Received unhandled message  
>> 125"
>> Any idea what that means? As far as I know there is no <external>  
>> result
>> code 125 defined in the ISDN protocol.
>>
>> Best regards
>> Peter
>>
>> See the following logs.
>>
>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14
>> --- 
>> --- 
>> --- 
>> --- 
>> --------------------------------------------------------------------
>> [08 02 80 03 7d 08 04 82 e3 98 28 14 01 01]
>>
>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Received message from Q.921
>> (ind 4, tei 0, size 18)
>> MesType: 125, CRVFlag 1 (Terminator), CRV 3 (Dialect: 2)
>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Sending message to Layer4
>> (size: 114)
>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I  
>> got
>> an event! Type:[7d] Size:[114] CRV: 3 (0x3, CTX: Terminator)
>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan
>> 70d230 (1:1) source isdn_data->channels_local_crv[0x3]
>> 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received
>> unhandled message 125 (0x7d)
>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.921() Sending frame
>>
>>
>>> oz list
>> API CALL [oz(list)] output:
>> +OK
>> span: 1 (span1)
>> type: isdn
>> chan_count: 31
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>> +OK
>> span: 2 (span2)
>> type: isdn
>> chan_count: 31
>> dialplan: XML
>> context: default
>> dial_regex:
>> fail_dial_regex:
>> hold_music:
>> analog_options none
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
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WKkAnjkDOXKrgFKQNHyeSyvw6rI2JW0U
=Ax8Y
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From fidibus83 at aol.com  Tue Dec 16 06:49:17 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 16 Dec 2008 15:49:17 +0100
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org>
References: <002101c95f64$56958e60$6445310a@Franzi><7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com><007601c95f85$c1462aa0$6445310a@Franzi>
	<9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org>
Message-ID: <009f01c95f8d$7918ddb0$6445310a@Franzi>

It?s already the same error.

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael S Collins
Gesendet: Dienstag, 16. Dezember 2008 15:37
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

Just a hunch but try removing the spaces in this line:

 

loadzone=de

Zaptel can be quirky. 

-MC


Sent from my iPhone


On Dec 16, 2008, at 5:54 AM, "fidibus83" <fidibus83 at aol.com> wrote:

I have installed zaptel-1.4.11 

I have looked in zonedata.c and there is configured de-tonezone 

 


  _____  


Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael Jerris
Gesendet: Dienstag, 16. Dezember 2008 14:20
An:  <mailto:freeswitch-users at lists.freeswitch.org>
freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

It sounds like there is no de tonezone in the zaptel drivers, but I can't
imagine thats true.  What version of the drivers do you have installed?  

 

 

On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:








Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

 

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg ?vv:

 

31 channels configured.

 

ioctl(ZT_LOADZONE) failed: Invalid argument

Notice: Configuration file is /etc/zaptel.conf

line 288: Unable to register tone zone 'de'

 

 

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

What do I wrong? Please help me!

_______________________________________________
Freeswitch-users mailing list
 <mailto:Freeswitch-users at lists.freeswitch.org>
Freeswitch-users at lists.freeswitch.org
 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>
http://lists.freeswitch.org/mailman/options/freeswitch-users
 <http://www.freeswitch.org> http://www.freeswitch.org

 

= 

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>
http://lists.freeswitch.org/mailman/options/freeswitch-users
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=
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From helmut.kuper at ewetel.de  Tue Dec 16 06:58:21 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 15:58:21 +0100
Subject: [Freeswitch-users] mod_ldap
In-Reply-To: <818373FA-A5DE-4534-B807-CDE119E9EE42@jerris.com>
References: <4947A7F3.90703@ewetel.de>
	<818373FA-A5DE-4534-B807-CDE119E9EE42@jerris.com>
Message-ID: <4947C20D.4010704@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

k did it: http://jira.freeswitch.org/browse/MDXMLINT-44

Didn't know where to put it, so I put it to "XML interfaces"

regards
helmut


Am 16.12.2008 14:32, schrieb Michael Jerris:
> Can you please file a bug in jira on this issue, we can get this  
> corrected today.
> On Dec 16, 2008, at 8:06 AM, Helmut Kuper wrote:
>
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From mike at jerris.com  Tue Dec 16 07:00:31 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 10:00:31 -0500
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <4947BC9C.8040302@ewetel.de>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>
	<49479FD7.2070200@ewetel.de>
	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>
	<4947BC9C.8040302@ewetel.de>
Message-ID: <E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>

possibly you have older code that was not bootstrapped again when we  
added spandsp?  Try update, bootstrap, configure again and see if its  
there after.

Mike

On Dec 16, 2008, at 9:35 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> hm no, there is nothing like libtiff, spandsp or even fax to find in
> config.log.
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklHvJsACgkQ4tZeNddg3dwsOwCeI1hVTC3CeW/SLiuHn0g6VTYW
> 1WgAn1dOjzoBgU8Ln6Wri/a53O8rRZOO
> =fHh+
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Tue Dec 16 07:02:10 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 16 Dec 2008 10:02:10 -0500
Subject: [Freeswitch-users] OpenZAP:  "Received unhandled message 125"
In-Reply-To: <4947BF0E.50702@ewetel.de>
References: <492ED886.9020805@gmx.net>
	<EBD35DE7-A6C9-4072-AC8E-F2D49E7098C8@freeswitch.org>
	<4947BF0E.50702@ewetel.de>
Message-ID: <8521EE9F-DBE2-40B5-B188-FE124DA77002@jerris.com>

That is just a message we don't handle yet, it will be added in the  
future.

Mike

On Dec 16, 2008, at 9:45 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I get that "CRIT" error on my q931 pri as well. I found that the  
> message
> is a Q931 status message from remote end and I guess it only will tell
> you the call state of your outgoing call at the remote end to make  
> sure
> you are in same state. So it is a nice to have to get this mesage
> decoded for debugging reasons.
>>> 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34()  
>>> Received
>>> unhandled message 125 (0x7d)
>>>



From vkobashi at ydeasolutions.com.br  Tue Dec 16 07:08:24 2008
From: vkobashi at ydeasolutions.com.br (Vinicius Kobashi)
Date: Tue, 16 Dec 2008 13:08:24 -0200
Subject: [Freeswitch-users] mod_ldap
In-Reply-To: <4947A7F3.90703@ewetel.de>
References: <4947A7F3.90703@ewetel.de>
Message-ID: <4947C468.7050100@ydeasolutions.com.br>

yes it is, you need to hack fs to get the latest version of openldap for you

Helmut Kuper escreveu:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I updated to latest FS trunk today and tried to compile mod_ldap and got
> this errors:
>
>
> mkdir .libs
> ar cru .libs/libldap.a  bind.o open.o result.o error.o compare.o
> search.o controls.o messages.o references.o extended.o cyrus.o modify.o
> add.o modrdn.o delete.o abandon.o sasl.o sbind.o kbind.o unbind.o
> cancel.o filter.o free.o sort.o passwd.o whoami.o getdn.o getentry.o
> getattr.o getvalues.o addentry.o request.o os-ip.o url.o sortctrl.o
> vlvctrl.o init.o options.o print.o string.o util-int.o schema.o
> charray.o tls.o os-local.o dnssrv.o utf-8.o utf-8-conv.o turn.o
> groupings.o txn.o ppolicy.o version.o
> ranlib .libs/libldap.a
> creating libldap.la
> (cd .libs && rm -f libldap.la && ln -s ../libldap.la libldap.la)
> cc -g -O2 -o apitest apitest.o  ./.libs/libldap.a
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/liblber/.libs/liblber.a
> ../../libraries/liblber/.libs/liblber.a
> ../../libraries/liblutil/liblutil.a /usr/lib/libsasl2.a -ldl -lresolv
> ./.libs/libldap.a(os-ip.o): In function `ldap_pvt_is_socket_ready':
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194:
> warning: `sys_errlist' is deprecated; use `strerror' or `strerror_r' instead
> /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194:
> warning: `sys_nerr' is deprecated; use `strerror' or `strerror_r' instead
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_close':
> (.text+0x348): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
> (.text+0x3d0): undefined reference to `db_create'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open':
> (.text+0x453): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_getdata':
> (.text+0x7e8): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
> (.text+0xa4e): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata':
> (.text+0xafa): undefined reference to `db_strerror'
> /usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
> (.text+0x6c8): undefined reference to `DES_key_sched'
> /usr/lib/libsasl2.a(digestmd5.o): In function `init_des':
>
>
> When I change src/mod/directories/mod_ldap/Makefile line 3 to
> "LDAP=openldap-2.4.11" and putting openldap-2.4.11.tgz in libs/ manually
> it compiles without error, but with some warnings.
>
>
> Maybe the openldap-2.3.19 delivered by FS is too old?
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
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> 25kAoK0igc53iV1PhWkj5faAVOL78E8B
> =ny1E
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From anthony.minessale at gmail.com  Tue Dec 16 07:17:29 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Dec 2008 09:17:29 -0600
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <21033993.post@talk.nabble.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
Message-ID: <191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>

it should work with
<action application="bind_meta_app" data="1 ab s transfer::-both 3333"/>

make sure you have debug log on (press f8) to see if there are any issues.


On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> Thanks for this answer.
> Just a question so that everything is clear: there is no command to bridge
> directly a call and both legs into a conference? We have to transfer the
> call to another extension and from there create the conference isn't?
>
> I used the command transfer but I still have a problem. I do the following
> and it did not work: in my dialplan I write:
> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
>
> The extension 3333 works fine.
>
> However, nothing happens when I press *1.
> Do you have an idea where am I wrong?
>
> (A subsidiary question: will both legs be transferred to the extension 3333
> or just the one which press *1? is there a way to transfer both together?)
>
>
> Thanks a lot,
> Carole
>
>
> Anthony Minessale-2 wrote:
> >
> > There is a transfer dial plan application also.
> >
> > There is not really any benefit in blocking the api commands from the
> > dialplan
> > apart from the potential for a blocking api call to delay the audio
> stream
> > which
> > you can do at your own risk and use the sleep application with 0ms to
> > flush
> > the read buffers.
> >
> > So I guess i can lift that limitation in tree.
> >
> > see r10790
> >
> >
> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
> wrote:
> >
> >>
> >> Hello,
> >>
> >> I have a general question about the API commands. Some of them are not
> >> available in the dialplan like uuid_transfer. I was wondering how to
> call
> >> an
> >> API command without using the CLI. Especially I would be interested in
> >> knowing if there is any way to call them from a phone, I mean bind a key
> >> to
> >> an API command.
> >>
> >> For instance I would like to transfer both members of a simple call into
> >> a
> >> conference by dialing *1.
> >> I have seen the transfer is possible from the CLI by doing:
> >> api uuid_transfer <uuid> -both 3001
> >> but I do not know how to do it else.
> >>
> >> If somebody could give me an insight about the topic it would be great.
> >> Thanks,
> >> Carole
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> --
> >> View this message in context:
> >>
> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From helmut.kuper at ewetel.de  Tue Dec 16 07:39:55 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 16:39:55 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
Message-ID: <4947CBCB.8060204@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

H Anthony,

I have same problem on each 7th outgoing call. Then a sort of timeout
occurs. During that time, no incomming call is possible.

regards
helmut


Am 23.09.2008 16:13, schrieb Anthony Minessale:
> I have moved your issue to  http://jira.freeswitch.org/browse/OPENZAP-18
> since it's an openzap issue and not a build system issue.
> 
> Please pay attention to the emails you receive regarding the issue and
> promptly reply to any comments.
> 
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Version: GnuPG v1.4.9 (MingW32)

iEYEARECAAYFAklHy8sACgkQ4tZeNddg3dynXgCeLco/4Pmu3hFVVFHP/glucTtS
o4oAoIo1q2bA5ItCwGx4y9kZsy1KPWlp
=Rv7D
-----END PGP SIGNATURE-----



From msc at freeswitch.org  Tue Dec 16 07:49:46 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Dec 2008 07:49:46 -0800
Subject: [Freeswitch-users] OpenZAP:  "Received unhandled message 125"
In-Reply-To: <4947BF0E.50702@ewetel.de>
References: <492ED886.9020805@gmx.net>
	<EBD35DE7-A6C9-4072-AC8E-F2D49E7098C8@freeswitch.org>
	<4947BF0E.50702@ewetel.de>
Message-ID: <CC1175B5-5B44-4C65-A836-D6079BA6F358@freeswitch.org>

It's on the list of things to do. Since there is no bounty for getting  
the pri stack finished it's a little bit lower priority. However, it  
is definitely being worked on.

-MC

Sent from my iPhone

On Dec 16, 2008, at 6:45 AM, Helmut Kuper <helmut.kuper at ewetel.de>  
wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> I get that "CRIT" error on my q931 pri as well. I found that the  
> message
> is a Q931 status message from remote end and I guess it only will tell
> you the call state of your outgoing call at the remote end to make  
> sure
> you are in same state. So it is a nice to have to get this mesage
> decoded for debugging reasons.
>
> regards
> helmut
>
>
>
> Am 27.11.2008 19:02, schrieb Michael S Collins:
>> You can ignore this one for now. Eventually this will be handled but
>> it shouldn't affect your calls. I've been ignoring it for six  
>> months. :)
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Nov 27, 2008, at 9:27 AM, Peter P GMX <Prometheus001 at gmx.net>  
>> wrote:
>>
>>> I have installed OpenZAP with a TE220 card and EuroISDN.
>>>
>>> When I bridge an outgoing call I get a "Received unhandled message
>>> 125"
>>> Any idea what that means? As far as I know there is no <external>
>>> result
>>> code 125 defined in the ISDN protocol.
>>>
>>> Best regards
>>> Peter
>>>
>>> See the following logs.
>>>
>>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23()  
>>> READ 14
>>> ---
>>> ---
>>> ---
>>> ---
>>> --------------------------------------------------------------------
>>> [08 02 80 03 7d 08 04 82 e3 98 28 14 01 01]
>>>
>>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Received message from Q. 
>>> 921
>>> (ind 4, tei 0, size 18)
>>> MesType: 125, CRVFlag 1 (Terminator), CRV 3 (Dialect: 2)
>>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Sending message to Layer4
>>> (size: 114)
>>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I
>>> got
>>> an event! Type:[7d] Size:[114] CRV: 3 (0x3, CTX: Terminator)
>>> 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan
>>> 70d230 (1:1) source isdn_data->channels_local_crv[0x3]
>>> 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34()  
>>> Received
>>> unhandled message 125 (0x7d)
>>> 2008-11-27 17:35:17 [DEBUG] Span:0 Q.921() Sending frame
>>>
>>>
>>>> oz list
>>> API CALL [oz(list)] output:
>>> +OK
>>> span: 1 (span1)
>>> type: isdn
>>> chan_count: 31
>>> dialplan: XML
>>> context: default
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options none
>>> +OK
>>> span: 2 (span2)
>>> type: isdn
>>> chan_count: 31
>>> dialplan: XML
>>> context: default
>>> dial_regex:
>>> fail_dial_regex:
>>> hold_music:
>>> analog_options none
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklHvw0ACgkQ4tZeNddg3dyAnACfaXoHklB3xfaaQXTB+raTyASV
> WKkAnjkDOXKrgFKQNHyeSyvw6rI2JW0U
> =Ax8Y
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Tue Dec 16 07:52:41 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Dec 2008 07:52:41 -0800
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <009f01c95f8d$7918ddb0$6445310a@Franzi>
References: <002101c95f64$56958e60$6445310a@Franzi><7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com><007601c95f85$c1462aa0$6445310a@Franzi>
	<9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org>
	<009f01c95f8d$7918ddb0$6445310a@Franzi>
Message-ID: <B4F10400-D789-4507-8C06-F2B7F79ECED7@freeswitch.org>

Thanks for trying. You might want to clean out your zaptel install and  
do a complete reinstall.

-MC

Sent from my iPhone

On Dec 16, 2008, at 6:49 AM, "fidibus83" <fidibus83 at aol.com> wrote:

> It?s already the same error.
>
>
>
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] Im Auftrag von Michael S Collins
> Gesendet: Dienstag, 16. Dezember 2008 15:37
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Zaptel Error!!!
>
>
>
> Just a hunch but try removing the spaces in this line:
>
>
>
> loadzone=de
>
> Zaptel can be quirky.
>
> -MC
>
>
> Sent from my iPhone
>
>
> On Dec 16, 2008, at 5:54 AM, "fidibus83" <fidibus83 at aol.com> wrote:
>
>> I have installed zaptel-1.4.11
>>
>> I have looked in zonedata.c and there is configured de-tonezone
>>
>>
>>
>> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
>> ] Im Auftrag von Michael Jerris
>> Gesendet: Dienstag, 16. Dezember 2008 14:20
>> An: freeswitch-users at lists.freeswitch.org
>> Betreff: Re: [Freeswitch-users] Zaptel Error!!!
>>
>>
>>
>> It sounds like there is no de tonezone in the zaptel drivers, but I  
>> can't imagine thats true.  What version of the drivers do you have  
>> installed?
>>
>>
>>
>>
>>
>> On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:
>>
>>
>>
>>
>>
>> Hello, I?m a newbie in FS and my English isn?t very good but I  
>> try to explain my problem. Hopefully you can understand me. J
>>
>>
>>
>> I have a Linux-Server with a Digium Wildcard TE110P. I install and  
>> configure openzap (PRI/ISDN) and zaptel. But I have an error when I  
>> execute ztcfg ?vv:
>>
>>
>>
>> 31 channels configured.
>>
>>
>>
>> ioctl(ZT_LOADZONE) failed: Invalid argument
>>
>> Notice: Configuration file is /etc/zaptel.conf
>>
>> line 288: Unable to register tone zone 'de'
>>
>>
>>
>>
>>
>>
>>
>> ZAPTEL.CONF
>>
>>
>>
>> span=1,1,0,ccs,hdb3,crc4
>>
>> bchan=1-15,17-31
>>
>> dchan=16
>>
>> loadzone = de
>>
>> defaultzone=de
>>
>>
>>
>> What do I wrong? Please help me!
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> =
>>
>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> = h.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> = y> =
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Tue Dec 16 08:04:32 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Dec 2008 08:04:32 -0800
Subject: [Freeswitch-users] Speed Dial Emulation
Message-ID: <686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>

Do you need something just for one extension? Or system wide?

If it's system wide then all you need is an extension that matches a  
condition like this:

<condition field="destination_number" expression="^1$">

Anyone who dials just a single digit 1 will go to this Dialplan entry.  
 From there you can have it bridge to whatever endpoint you like.

-MC

Sent from my iPhone

On Dec 15, 2008, at 7:45 AM, jflowers <jflowers at ezo.net> wrote:

>
> How do I emulate a speed dial setup.  That is, from extension 1003 I  
> dial
> just a 1 ( or 2, or 3 etc.) and nothing else and freeswitch dials a  
> PSTN
> number.  Is there software to do this?
>
>
> -- 
> View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21016167.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Tue Dec 16 08:10:42 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 10:10:42 -0600
Subject: [Freeswitch-users] Speed Dial Emulation
In-Reply-To: <686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>
References: <686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>
Message-ID: <154FFEA4-3E17-4570-B8D6-E54ED2D4A028@freeswitch.org>

Tie that to the db application and you'll have it.

/b

On Dec 16, 2008, at 10:04 AM, Michael S Collins wrote:

> Do you need something just for one extension? Or system wide?
>
> If it's system wide then all you need is an extension that matches a
> condition like this:
>
> <condition field="destination_number" expression="^1$">
>
> Anyone who dials just a single digit 1 will go to this Dialplan entry.
> From there you can have it bridge to whatever endpoint you like.
>
> -MC
>
> Sent from my iPhone




From helmut.kuper at ewetel.de  Tue Dec 16 08:44:17 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 17:44:17 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <4947CBCB.8060204@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de>
Message-ID: <4947DAE1.1050706@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello again,

hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
about  network congestion while mine is about a pri blocking timeout on
every 7th outgoing call.
So, shall I open a seperate bug on jira?

regards
helmut



Am 16.12.2008 16:39, schrieb Helmut Kuper:
> H Anthony,
> 
> I have same problem on each 7th outgoing call. Then a sort of timeout
> occurs. During that time, no incomming call is possible.
> 
> regards
> helmut
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0ugAnj1kX0w1wI45zjeouVz7VMA3d1UN
=H2ic
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From msc at freeswitch.org  Tue Dec 16 09:00:11 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Dec 2008 09:00:11 -0800
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <4947DAE1.1050706@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
Message-ID: <87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>

Yes please. Also, if you can attach some debug information, preferably
reproducing the symptoms with the full debug turned on (F8) and attach
it as a file that would assist with the research.

-MC

On Tue, Dec 16, 2008 at 8:44 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello again,
>
> hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
> about  network congestion while mine is about a pri blocking timeout on
> every 7th outgoing call.
> So, shall I open a seperate bug on jira?
>
> regards
> helmut
>
>
>
> Am 16.12.2008 16:39, schrieb Helmut Kuper:
>> H Anthony,
>>
>> I have same problem on each 7th outgoing call. Then a sort of timeout
>> occurs. During that time, no incomming call is possible.
>>
>> regards
>> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklH2uEACgkQ4tZeNddg3dxZ1ACgs5wZKty2kvzbqa57NONhJ67R
> 0ugAnj1kX0w1wI45zjeouVz7VMA3d1UN
> =H2ic
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From helmut.kuper at ewetel.de  Tue Dec 16 08:59:50 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 17:59:50 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>	<49479FD7.2070200@ewetel.de>	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>	<4947BC9C.8040302@ewetel.de>
	<E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>
Message-ID: <4947DE86.2010904@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Mike,

very likely. Currently I update an recompile FS serveral times a day :/

I did a "svn update", deleted the libs directory, did a bootstrap and
configure. This time in config.log I have a hint about spandsp :)

Buuuuuut ... I get this now :

making install mod_fax
Compiling mod_fax.c...
Creating mod_fax.so...
quiet_libtool: link: cannot find the library
`../../../../libs/spandsp/src/libspandsp.la' or unhandled argument
`../../../../libs/spandsp/src/libspandsp.la'
make[5]: *** [mod_fax.so] Error 1
make[4]: *** [install] Error 1
make[3]: *** [mod_fax-install] Error 1
make[2]: *** [install-recursive] Error 1
Making install in build

This time I had only to do a "make" in libs/spandsp directory to get FS
compiled.


regards
Helmut

Am 16.12.2008 16:00, schrieb Michael Jerris:
> possibly you have older code that was not bootstrapped again when we  
> added spandsp?  Try update, bootstrap, configure again and see if its  
> there after.
> 
> Mike
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=ahti
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From brian at freeswitch.org  Tue Dec 16 09:03:50 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 11:03:50 -0600
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <4947DE86.2010904@ewetel.de>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>	<49479FD7.2070200@ewetel.de>	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>	<4947BC9C.8040302@ewetel.de>
	<E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>
	<4947DE86.2010904@ewetel.de>
Message-ID: <D4E0568C-D219-4B3E-B964-F0C641DA805B@freeswitch.org>

You deleted the libs folder?  That might be why.

/b

On Dec 16, 2008, at 10:59 AM, Helmut Kuper wrote:

> I did a "svn update", deleted the libs directory, did a bootstrap and
> configure. This time in config.log I have a hint about spandsp :)




From helmut.kuper at ewetel.de  Tue Dec 16 09:14:46 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 18:14:46 +0100
Subject: [Freeswitch-users] FS mod_fax
In-Reply-To: <D4E0568C-D219-4B3E-B964-F0C641DA805B@freeswitch.org>
References: <493E435F.4010402@ewetel.de>	<FD1D1336-3531-4324-A4A0-5EC094F47EDA@jerris.com>	<493E9826.108@ewetel.de>	<87f2f3b90812090837o718d97ahb66d0854af0aee8b@mail.gmail.com>	<493F7148.40705@ewetel.de>	<49479FD7.2070200@ewetel.de>	<0201CA8A-388C-4C98-BC43-78C1CF52CBC0@jerris.com>	<4947BC9C.8040302@ewetel.de>	<E7A97D7E-40CC-4CB4-895A-C2AF9E5F0F02@jerris.com>	<4947DE86.2010904@ewetel.de>
	<D4E0568C-D219-4B3E-B964-F0C641DA805B@freeswitch.org>
Message-ID: <4947E206.7080504@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

oh sorry m8. first I deleted libs dir, then I did a svn update, so ne
fresh libs dir is installed  ... it's late ...  sorry


helmut


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From anthony.minessale at gmail.com  Tue Dec 16 09:35:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Dec 2008 11:35:51 -0600
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
Message-ID: <191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>

It looks like the other side is not responding at all to the setup message
so the span is locked waiting to see which channel should be assigned.
There is a mutex here stopping the span from using any more channels until
it gets a reply to that setup message.

We could either lower the 60 second timeout to a smaller val, unlock the
mutex before we make the call (possible race condition) or figure out why
you are getting no reply at all from your setup message.


On Tue, Dec 16, 2008 at 11:00 AM, Michael Collins <msc at freeswitch.org>wrote:

> Yes please. Also, if you can attach some debug information, preferably
> reproducing the symptoms with the full debug turned on (F8) and attach
> it as a file that would assist with the research.
>
> -MC
>
> On Tue, Dec 16, 2008 at 8:44 AM, Helmut Kuper <helmut.kuper at ewetel.de>
> wrote:
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: SHA1
> >
> > Hello again,
> >
> > hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
> > about  network congestion while mine is about a pri blocking timeout on
> > every 7th outgoing call.
> > So, shall I open a seperate bug on jira?
> >
> > regards
> > helmut
> >
> >
> >
> > Am 16.12.2008 16:39, schrieb Helmut Kuper:
> >> H Anthony,
> >>
> >> I have same problem on each 7th outgoing call. Then a sort of timeout
> >> occurs. During that time, no incomming call is possible.
> >>
> >> regards
> >> helmut
> > -----BEGIN PGP SIGNATURE-----
> > Version: GnuPG v1.4.9 (MingW32)
> >
> > iEYEARECAAYFAklH2uEACgkQ4tZeNddg3dxZ1ACgs5wZKty2kvzbqa57NONhJ67R
> > 0ugAnj1kX0w1wI45zjeouVz7VMA3d1UN
> > =H2ic
> > -----END PGP SIGNATURE-----
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From helmut.kuper at ewetel.de  Tue Dec 16 09:56:02 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 16 Dec 2008 18:56:02 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>	<4947CBCB.8060204@ewetel.de>
	<4947DAE1.1050706@ewetel.de>	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
Message-ID: <4947EBB2.7030401@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

yes I agree, but hey, the other side is a matured EWSD unlikely that
there is an issue left on protocol level after 30 years. I check that
tomorrow.

regards
helmut

Am 16.12.2008 18:35, schrieb Anthony Minessale:
> It looks like the other side is not responding at all to the setup message
> so the span is locked waiting to see which channel should be assigned.
> There is a mutex here stopping the span from using any more channels until
> it gets a reply to that setup message.
> 
> We could either lower the 60 second timeout to a smaller val, unlock the
> mutex before we make the call (possible race condition) or figure out why
> you are getting no reply at all from your setup message.
> 
> 
> On Tue, Dec 16, 2008 at 11:00 AM, Michael Collins <msc at freeswitch.org>wrote:
> 
>> Yes please. Also, if you can attach some debug information, preferably
>> reproducing the symptoms with the full debug turned on (F8) and attach
>> it as a file that would assist with the research.
>>
>> -MC
>>
>> On Tue, Dec 16, 2008 at 8:44 AM, Helmut Kuper <helmut.kuper at ewetel.de>
>> wrote:
> Hello again,
> 
> hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
> about  network congestion while mine is about a pri blocking timeout on
> every 7th outgoing call.
> So, shall I open a seperate bug on jira?
> 
> regards
> helmut
> 
> 
> 
> Am 16.12.2008 16:39, schrieb Helmut Kuper:
>>>>> H Anthony,
>>>>>
>>>>> I have same problem on each 7th outgoing call. Then a sort of timeout
>>>>> occurs. During that time, no incomming call is possible.
>>>>>
>>>>> regards
>>>>> helmut
>>>
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
>>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>

> ------------------------------------------------------------------------

> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Tue Dec 16 10:06:56 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Dec 2008 12:06:56 -0600
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <4947EBB2.7030401@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
	<4947EBB2.7030401@ewetel.de>
Message-ID: <191c3a030812161006s3ceac738m184304f47ca7eb87@mail.gmail.com>

yes I doubt your EWSD is doing anything wrong, but the communication channel
must be interrupted somehow to never get any replies so yes we can try to
figure out what has gone wrong in the communication layer.



On Tue, Dec 16, 2008 at 11:56 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> yes I agree, but hey, the other side is a matured EWSD unlikely that
> there is an issue left on protocol level after 30 years. I check that
> tomorrow.
>
> regards
> helmut
>
> Am 16.12.2008 18:35, schrieb Anthony Minessale:
> > It looks like the other side is not responding at all to the setup
> message
> > so the span is locked waiting to see which channel should be assigned.
> > There is a mutex here stopping the span from using any more channels
> until
> > it gets a reply to that setup message.
> >
> > We could either lower the 60 second timeout to a smaller val, unlock the
> > mutex before we make the call (possible race condition) or figure out why
> > you are getting no reply at all from your setup message.
> >
> >
> > On Tue, Dec 16, 2008 at 11:00 AM, Michael Collins <msc at freeswitch.org
> >wrote:
> >
> >> Yes please. Also, if you can attach some debug information, preferably
> >> reproducing the symptoms with the full debug turned on (F8) and attach
> >> it as a file that would assist with the research.
> >>
> >> -MC
> >>
> >> On Tue, Dec 16, 2008 at 8:44 AM, Helmut Kuper <helmut.kuper at ewetel.de>
> >> wrote:
> > Hello again,
> >
> > hm the bug OPENZAP-18 is not really the same problem as I have. OZ-18 is
> > about  network congestion while mine is about a pri blocking timeout on
> > every 7th outgoing call.
> > So, shall I open a seperate bug on jira?
> >
> > regards
> > helmut
> >
> >
> >
> > Am 16.12.2008 16:39, schrieb Helmut Kuper:
> >>>>> H Anthony,
> >>>>>
> >>>>> I have same problem on each 7th outgoing call. Then a sort of timeout
> >>>>> occurs. During that time, no incomming call is possible.
> >>>>>
> >>>>> regards
> >>>>> helmut
> >>>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> >>>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
>
> > ------------------------------------------------------------------------
>
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
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>
> iEYEARECAAYFAklH67IACgkQ4tZeNddg3dyuyQCeJYjTLdCCWjE3pIrFnwbw/cY2
> ZP4AoIFVFdWEPuIcy0Lbi5yhZ4/4sT/5
> =Q1vB
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Dec 16 10:13:57 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 12:13:57 -0600
Subject: [Freeswitch-users] FreeSWITCH Infrastructure / Happy Holidays!
Message-ID: <80B00891-6226-44DE-9430-6C3CC2A22D3F@freeswitch.org>

FreeSWITCers,
	As the project grows I felt the need for the project to own the  
critical infrastructure hosting things like SVN, Jira, Fisheye and  
various other things we as a project use.  I have personally paid for  
two servers that Bandwidth.com is going to be colocating for the  
project in their data center.  Teliax paid for and shipped me the two  
3ware raid1 cards for the machines which was a great help.  The  
servers are going into production after the first of the year and in  
the process we are rebuilding everything to allow for single sign on  
for every service on the FreeSWITCH project.

The servers are in the hands of bandwidth.com to be racked up so we  
can start deployment, as you can see we have our work cut out for us  
with the integration and rebuild of our entire infrastructure.  ;)

If you wish to pitch in my personal paypal is brian at freeswitch.org or  
you can use chipin http://www.chipin.com/contribute/id/ddc094318ae5ab5b

I would like to thank Bandwidth.com and Teliax for their support... I  
hope others will help out!

Also we have another sounds order going in sometime near the end of  
January so any money that exceeds the server cost will go to that.   
The sound order I placed monday will be in this Friday.

Happy Holidays and wish you the best for 2009!

Thanks,
Brian West
FreeSWITCH





From jflowers at ezo.net  Tue Dec 16 11:56:46 2008
From: jflowers at ezo.net (jflowers)
Date: Tue, 16 Dec 2008 11:56:46 -0800 (PST)
Subject: [Freeswitch-users] Speed Dial Emulation
In-Reply-To: <154FFEA4-3E17-4570-B8D6-E54ED2D4A028@freeswitch.org>
References: <21016167.post@talk.nabble.com>
	<686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>
	<154FFEA4-3E17-4570-B8D6-E54ED2D4A028@freeswitch.org>
Message-ID: <21040508.post@talk.nabble.com>


What db application (hint hint)?;-)

So maybe what I have to do is "^\d$" and lookup $1 in a db to find the
number to dial?


Brian West-3 wrote:
> 
> Tie that to the db application and you'll have it.
-- 
View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21040508.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Tue Dec 16 12:02:07 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 14:02:07 -0600
Subject: [Freeswitch-users] Speed Dial Emulation
In-Reply-To: <21040508.post@talk.nabble.com>
References: <21016167.post@talk.nabble.com>
	<686336B5-F7E4-45AD-9409-EB20B14877B9@freeswitch.org>
	<154FFEA4-3E17-4570-B8D6-E54ED2D4A028@freeswitch.org>
	<21040508.post@talk.nabble.com>
Message-ID: <B0486A14-661C-4D7F-87A4-F80B0585DDB5@freeswitch.org>

Its called "db" :P  check the default dialplan for the insert and  
select examples

/b

On Dec 16, 2008, at 1:56 PM, jflowers wrote:

>
> What db application (hint hint)?;-)
>
> So maybe what I have to do is "^\d$" and lookup $1 in a db to find the
> number to dial?




From jlists at skopis.com  Tue Dec 16 17:50:09 2008
From: jlists at skopis.com (John Skopis (Lists))
Date: Tue, 16 Dec 2008 19:50:09 -0600
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <4947A61F.6060806@ydeasolutions.com.br>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>	<494198F3.10806@ydeasolutions.com.br>
	<4947A61F.6060806@ydeasolutions.com.br>
Message-ID: <49485AD1.5070708@skopis.com>

vinicius wrote:
> hi ppl.. i tried to find something at google, but i couldnt manage to find
> anything.
> i still dont know what to do to make the mod_xml_ldap work.
> i couldnt find information about how to build a config file for the
> module, and where to store it...
> 
> can anyone give me a help?
> 

Be advised mod_xml_ldap is probably not production quality and will
undoubtedly change, eventually at least.

Here is what I used once:

  <bindings>


    <binding name="directory">
		<!--%s is populated with the extension -->
		<param name="filter" value="(FSid=%s)" bindings="directory"/>
		<!--basedn for the searches %s is replaced with domain-->
		<param name="basedn" value="ou=people,dc=example" />
		<param name="url" value="ldap://172.16.75.129" />
		<param name="binddn" value="cn=admin,dc=example" />
		<param name="bindpass" value="secret" />

		<trans>
			<!-- we need to translate these attrs into FS attrs -->
			<tran name="id" mapfrom="FSid" />
			<tran name="mailbox" mapfrom="FSmailbox" />
			<tran name="password" mapfrom="FSPassword" />
			<tran name="vm-password" mapfrom="FSvm-password" />
			<tran name="email-addr" mapfrom="FSemail-addr" />
			<tran name="vm-email-all-messages" mapfrom="FSvm-email-all-messages" />
			<tran name="vm-delete-file" mapfrom="FSvm-delete-file" />
			<tran name="vm-attach-file" mapfrom="FSvm-attach-file" />
		</trans>
    </binding>

    <binding name="configuration">
        <param name="filter" value="(%s=%s)" bindings="configuration"/>
        <param name="basedn" value="name=%s,dc=example" />
        <param name="url" value="ldap://172.16.75.129" />
        <param name="binddn" value="cn=admin,dc=example" />
        <param name="bindpass" value="secret" />
    </binding>
</bindings>


which should/probably/might work with ldap objects like these:

dn: cn=John Skopis,ou=people,dc=example
objectClass: person
objectClass: inetOrgPerson
objectClass: organizationalPerson
objectClass: FreeSWITCH-Exten-Object
objectClass: top
cn: John Skopis
sn: Skopis
givenName: John
FSid: 1001
FSmailbox: 1001
FSpassword: 1234
FSvm-password: 1001
FSemail-addr: john+fs at skopis.com
FSvm-email-all-messages: TRUE
FSvm-delete-file: TRUE
FSvm-attach-file: TRUE

dn: SIPIdentityUserName=1001,ou=h350,dc=example
objectClass: person
objectClass: SIPIdentity
objectClass: top
cn: 1001
sn: 1001
SIPIdentitySIPURI: sip:1001 at 172.16.75.129
SIPIdentityRegistrarAddress: 172.16.75.128
SIPIdentityProxyAddress: 172.16.75.128
SIPIdentityPassword: 1234
SIPIdentityUserName: 1001
SIPIdentityServiceLevel: premium




From marc at kasteris.com  Tue Dec 16 20:24:09 2008
From: marc at kasteris.com (Marc Orenberg)
Date: Tue, 16 Dec 2008 20:24:09 -0800 (PST)
Subject: [Freeswitch-users] Ending a bridged call with a touchtone
Message-ID: <52400.8491.qm@web50803.mail.re2.yahoo.com>

Hello.? I'm trying to allow the A-leg of a bridged call to be able to press a touchtone to end the call.
In my Python script, I set-up a DTMF callback function using setInputCallback, but it doesn't seem to?have any effect?during bridged calls. Is there another way to do this? Thanks.
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From brian at freeswitch.org  Tue Dec 16 09:30:46 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 11:30:46 -0600
Subject: [Freeswitch-users] Ending a bridged call with a touchtone
In-Reply-To: <52400.8491.qm@web50803.mail.re2.yahoo.com>
References: <52400.8491.qm@web50803.mail.re2.yahoo.com>
Message-ID: <F9E12702-5CA0-45A8-B408-3F2162F94A8E@freeswitch.org>

Try bind_meta, examples are in the default dialplan.

/b

On Dec 16, 2008, at 10:24 PM, Marc Orenberg wrote:

> Hello.  I'm trying to allow the A-leg of a bridged call to be able  
> to press a touchtone to end the call.
> In my Python script, I set-up a DTMF callback function using  
> setInputCallback, but it doesn't seem to have any effect during  
> bridged calls. Is there another way to do this? Thanks.

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From jason at jasonjgw.net  Tue Dec 16 21:07:01 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 17 Dec 2008 16:07:01 +1100
Subject: [Freeswitch-users] error creating IPv6 profile when corresponding
	IPv4 port in use
Message-ID: <20081217050701.GA14997@jdc.jasonjgw.net>

Here's the scenario (FreeSWITCH revision 10725).

I have Asterisk listening on port 5060 under IPv4. This version of Asterisk
does not support IPv6, and netstat -6 -a | grep sip suggests that, as
expected, it isn't listening on port 5060 under IPv6.

If I set up FreeSWITCH profiles to listen on port 5070, for example, under
both IPv4 and IPv6 then it works fine.

However, if I create an IPv4 profile to listen on port 5070, and an IPv6
profile to listen on port 5060, Sofia fails with a "port already in use"
error, even though nothing should be using port 5060 under IPv6.

>From the logs:

tport_create(): 0x1b4d0e0
tport_bind_server(0x1b4d0e0) to */[2001:44b8:61::3b]:5060/sip
tport_bind_server(0x1b4d0e0): calling tport_listen for udp
tport_alloc_primary(0x1b4d0e0): new primary tport 0x1b4f010
tport_listen(0x1b4d0e0): bind(pf=10 udp/[2001:44b8:61::3b]:5060): Address already in use
tport_destroy(0x1b4d0e0)
nta: bind([2001:44b8:61::3b]:5060;transport=*): Address already in use
nua: initializing SIP stack failed

Obviously, I can easily use a different port, and hence this issue doesn't
affect me. However, if it's a bug, I think it should still go into the bug
list. I realize the problem might not be in Sofia or FreeSWITCH. Is Asterisk
really taking over the IPv6 port as well?

Environment: Debian Sid, kernel 2.6.26, x86_64 architecture.

Note: once I learn enough about FreeSWITCH to configure it properly and set it
up to meet my personal telephony needs, the Asterisk installation will be,
shall we say, redundant.




From brian at freeswitch.org  Tue Dec 16 21:10:19 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Dec 2008 23:10:19 -0600
Subject: [Freeswitch-users] error creating IPv6 profile when
	corresponding IPv4 port in use
In-Reply-To: <20081217050701.GA14997@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
Message-ID: <4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>

Well the OS reports its in use when we try to bind not much we can do  
about that ... what does netstat -na | grep 5060 show?

/b

On Dec 16, 2008, at 11:07 PM, Jason White wrote:

> nta: bind([2001:44b8:61::3b]:5060;transport=*): Address already in use




From jason at jasonjgw.net  Tue Dec 16 21:19:35 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 17 Dec 2008 16:19:35 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
Message-ID: <20081217051935.GA15357@jdc.jasonjgw.net>

On Tue, Dec 16, 2008 at 11:10:19PM -0600, Brian West wrote:
> Well the OS reports its in use when we try to bind not much we can do  
> about that ... what does netstat -na | grep 5060 show?

udp        0      0 0.0.0.0:5060            0.0.0.0:*

I'll take it, then, that this is a non-issue?




From jason at jasonjgw.net  Tue Dec 16 23:40:02 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 17 Dec 2008 18:40:02 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <20081217051935.GA15357@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
Message-ID: <20081217074002.GA16365@jdc.jasonjgw.net>

The code in bind6only_check in libs/sofia-sip/libsofia-sip-ua/tport/tport.c looks
correct to me, but I can't find where the result is tested (it's in
mr_bindv6only). When bind6only_check() is called in tport_bind_server(), the
return value isn't tested, and I'm having difficulty finding where it is used
- I'm interested in whether we're in fact attempting to bind only to the IPv6
  port and whether the logic is correct here.

When I find the time, I could rebuild with debug symbols and run it under gdb.




From mszlazak at aol.com  Wed Dec 17 00:06:49 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 17 Dec 2008 03:06:49 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
In-Reply-To: <957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com>
References: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>
	<957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com>
Message-ID: <8CB2E358C58A406-9B8-1883@WEBMAIL-MA13.sysops.aol.com>


 Hi Mike,

That does get the audio go between the softphone and the application (Voxeo's Prophecy ASR) "around" FreeSwitch but I would like the audio going "through" FreeSwitch. I plan to do something to it before passing it on.

Support from Voxeo had this to say about the "bypass media" setting and if you could add some more insight that would be much appreciated. Since this is all on one Windows XP machine they can't get the info from the pcap file and are requesting I set up freeswitch on another machine which I will do. I thought you may have some more input.

?

Mark,

This is great news, it certainly confirms our suspicions that freeswitch was not 
forwarding media to Prophecy, or if so, it was doing it on a different port then 
we specified to be listening on.

To address the lingering question in this thread, I don't believe we have a firm 
enough grasp on your deployment calls to understand whether free-switch need the 
RTP stream or not.  If FreeSwitch is intended in your deployment to act as a 
front end for call routing to terminate calls to Prophecy then there is no need 
for it to listen to media.

Of course, it will hold the SIP communication tether so that it remains aware of 
disconnect events, would be my assumption, I am sure freeswitch can verify this 
behavior.

In order for us to understand why this config change is required will need a 
wireshark trace, and with your stacked approach to have both Prophecy and 
freeswitch on the
 same box makes this impossible.  For troubleshooting, if you 
moved freeswitch to another server temporarily, this may offer some insight into 
this problem, with wireshark at our disposal.

Hope this helps!



 


 

-----Original Message-----
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, 16 Dec 2008 5:24 am
Subject: Re: [Freeswitch-users] Help with routing sound locally through FS









If its all local you can also just use:



http://wiki.freeswitch.org/wiki/Bypass_Media




If your still trying to figure it out it could be any number of things, but most relating to misconfigured endpoints or freeswitch, take a look at the sip trace and make sure everything is using the right ip addresses instead of using internal when they should be external or the other way around.




Mike




On Dec 16, 2008, at 5:02 AM, mszlazak at aol.com wrote:


I'm making a call internally from a soft phone to an extension that is suppose to bridge the call internally to another application on the same computer. The applications logs indicate that a connection was made but sound is not being passed back from the application through freeswitch to the softphone. There maybe an issue with rtp timing and associated ports but I'm very new at diagnosing this and fixing the problem.
 
 I've attached both a copy of the FS log and an associated pcap file.
 
 It's all on Windows XP. 
 
 Could someone please take a look.
 
 Thanks.
 




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=


 





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From carole.olivier at enst.fr  Wed Dec 17 02:06:39 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 02:06:39 -0800 (PST)
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
	<191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
Message-ID: <21050314.post@talk.nabble.com>


Thanks, this works fine.

But I try to use some other API commands and something goes wrong: I would
like to be able to use the API commands for the conference like lock,
unlock, say, etc... from the dialplan.

I try to add in my dialplan 
<action application="bind_meta_app" data="1 a s conference::conf1 lock"/>
but it did not work, I presse F8 and I have got:

2008-12-17 10:47:58 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes Processing
meta digit '2' [conference::conf1 lock]
2008-12-17 10:47:58 [DEBUG] switch_core_session.c:611
switch_core_session_queue_private_event() Kill
sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes [BREAK]

You told me that the API commands should not be blocked isn't it? So I do
not understand where I am wrong. Is "bind_meta_app" not supported in
conference?

I have also tried to write directly:
<action application="conference" data="conf1 lock"/>
which also has no effect.

Thanks for your help,
Carole


Anthony Minessale-2 wrote:
> 
> it should work with
> <action application="bind_meta_app" data="1 ab s transfer::-both 3333"/>
> 
> make sure you have debug log on (press f8) to see if there are any issues.
> 
> 
> On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr> wrote:
> 
>>
>> Thanks for this answer.
>> Just a question so that everything is clear: there is no command to
>> bridge
>> directly a call and both legs into a conference? We have to transfer the
>> call to another extension and from there create the conference isn't?
>>
>> I used the command transfer but I still have a problem. I do the
>> following
>> and it did not work: in my dialplan I write:
>> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
>>
>> The extension 3333 works fine.
>>
>> However, nothing happens when I press *1.
>> Do you have an idea where am I wrong?
>>
>> (A subsidiary question: will both legs be transferred to the extension
>> 3333
>> or just the one which press *1? is there a way to transfer both
>> together?)
>>
>>
>> Thanks a lot,
>> Carole
>>
>>
>> Anthony Minessale-2 wrote:
>> >
>> > There is a transfer dial plan application also.
>> >
>> > There is not really any benefit in blocking the api commands from the
>> > dialplan
>> > apart from the potential for a blocking api call to delay the audio
>> stream
>> > which
>> > you can do at your own risk and use the sleep application with 0ms to
>> > flush
>> > the read buffers.
>> >
>> > So I guess i can lift that limitation in tree.
>> >
>> > see r10790
>> >
>> >
>> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
>> wrote:
>> >
>> >>
>> >> Hello,
>> >>
>> >> I have a general question about the API commands. Some of them are not
>> >> available in the dialplan like uuid_transfer. I was wondering how to
>> call
>> >> an
>> >> API command without using the CLI. Especially I would be interested in
>> >> knowing if there is any way to call them from a phone, I mean bind a
>> key
>> >> to
>> >> an API command.
>> >>
>> >> For instance I would like to transfer both members of a simple call
>> into
>> >> a
>> >> conference by dialing *1.
>> >> I have seen the transfer is possible from the CLI by doing:
>> >> api uuid_transfer <uuid> -both 3001
>> >> but I do not know how to do it else.
>> >>
>> >> If somebody could give me an insight about the topic it would be
>> great.
>> >> Thanks,
>> >> Carole
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> --
>> >> View this message in context:
>> >>
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >>
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com><
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >
>> >
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org><
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> > iax:guest at conference.freeswitch.org/888
>> >
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21050314.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From carole.olivier at enst.fr  Wed Dec 17 02:15:51 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 02:15:51 -0800 (PST)
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <21050314.post@talk.nabble.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
	<191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
	<21050314.post@talk.nabble.com>
Message-ID: <21050436.post@talk.nabble.com>


(I have just read the post again, I have written <action
application="bind_meta_app" data="1 a s conference::conf1 lock"/> but I
meant <action application="bind_meta_app" data="2 a s conference::conf1
lock"/> please don't pay attention for that, I made the mistake when I have
copied it in the post, not in the configuration.)


Carole O. wrote:
> 
> Thanks, this works fine.
> 
> But I try to use some other API commands and something goes wrong: I would
> like to be able to use the API commands for the conference like lock,
> unlock, say, etc... from the dialplan.
> 
> I try to add in my dialplan 
> <action application="bind_meta_app" data="1 a s conference::conf1 lock"/>
> but it did not work, I presse F8 and I have got:
> 
> 2008-12-17 10:47:58 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
> sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes Processing
> meta digit '2' [conference::conf1 lock]
> 2008-12-17 10:47:58 [DEBUG] switch_core_session.c:611
> switch_core_session_queue_private_event() Kill
> sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes [BREAK]
> 
> You told me that the API commands should not be blocked isn't it? So I do
> not understand where I am wrong. Is "bind_meta_app" not supported in
> conference?
> 
> I have also tried to write directly:
> <action application="conference" data="conf1 lock"/>
> which also has no effect.
> 
> Thanks for your help,
> Carole
> 
> 
> Anthony Minessale-2 wrote:
>> 
>> it should work with
>> <action application="bind_meta_app" data="1 ab s transfer::-both 3333"/>
>> 
>> make sure you have debug log on (press f8) to see if there are any
>> issues.
>> 
>> 
>> On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr>
>> wrote:
>> 
>>>
>>> Thanks for this answer.
>>> Just a question so that everything is clear: there is no command to
>>> bridge
>>> directly a call and both legs into a conference? We have to transfer the
>>> call to another extension and from there create the conference isn't?
>>>
>>> I used the command transfer but I still have a problem. I do the
>>> following
>>> and it did not work: in my dialplan I write:
>>> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
>>>
>>> The extension 3333 works fine.
>>>
>>> However, nothing happens when I press *1.
>>> Do you have an idea where am I wrong?
>>>
>>> (A subsidiary question: will both legs be transferred to the extension
>>> 3333
>>> or just the one which press *1? is there a way to transfer both
>>> together?)
>>>
>>>
>>> Thanks a lot,
>>> Carole
>>>
>>>
>>> Anthony Minessale-2 wrote:
>>> >
>>> > There is a transfer dial plan application also.
>>> >
>>> > There is not really any benefit in blocking the api commands from the
>>> > dialplan
>>> > apart from the potential for a blocking api call to delay the audio
>>> stream
>>> > which
>>> > you can do at your own risk and use the sleep application with 0ms to
>>> > flush
>>> > the read buffers.
>>> >
>>> > So I guess i can lift that limitation in tree.
>>> >
>>> > see r10790
>>> >
>>> >
>>> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
>>> wrote:
>>> >
>>> >>
>>> >> Hello,
>>> >>
>>> >> I have a general question about the API commands. Some of them are
>>> not
>>> >> available in the dialplan like uuid_transfer. I was wondering how to
>>> call
>>> >> an
>>> >> API command without using the CLI. Especially I would be interested
>>> in
>>> >> knowing if there is any way to call them from a phone, I mean bind a
>>> key
>>> >> to
>>> >> an API command.
>>> >>
>>> >> For instance I would like to transfer both members of a simple call
>>> into
>>> >> a
>>> >> conference by dialing *1.
>>> >> I have seen the transfer is possible from the CLI by doing:
>>> >> api uuid_transfer <uuid> -both 3001
>>> >> but I do not know how to do it else.
>>> >>
>>> >> If somebody could give me an insight about the topic it would be
>>> great.
>>> >> Thanks,
>>> >> Carole
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> View this message in context:
>>> >>
>>> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
>>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> Freeswitch-users mailing list
>>> >> Freeswitch-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> <MSN%3Aanthony_minessale at hotmail.com><
>>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>>> >
>>> >
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>>> >
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> <sip%3A888 at conference.freeswitch.org><
>>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>>> >
>>> > iax:guest at conference.freeswitch.org/888
>>> >
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>>> >
>>> > pstn:213-799-1400
>>> >
>>> > _______________________________________________
>>> > Freeswitch-users mailing list
>>> > Freeswitch-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 

-- 
View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21050436.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From fidibus83 at aol.com  Wed Dec 17 02:17:41 2008
From: fidibus83 at aol.com (fidibus83)
Date: Wed, 17 Dec 2008 11:17:41 +0100
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <B4F10400-D789-4507-8C06-F2B7F79ECED7@freeswitch.org>
References: <002101c95f64$56958e60$6445310a@Franzi><7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com><007601c95f85$c1462aa0$6445310a@Franzi><9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org><009f01c95f8d$7918ddb0$6445310a@Franzi>
	<B4F10400-D789-4507-8C06-F2B7F79ECED7@freeswitch.org>
Message-ID: <004201c96030$b2b9fd80$6445310a@Franzi>

I did a reinstall but there is the same error!

Is there something else I can do to remove the error?

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael S Collins
Gesendet: Dienstag, 16. Dezember 2008 16:53
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

Thanks for trying. You might want to clean out your zaptel install and do a
complete reinstall. 

 

-MC

Sent from my iPhone


On Dec 16, 2008, at 6:49 AM, "fidibus83" <fidibus83 at aol.com> wrote:

It?s already the same error.

 


  _____  


Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael S Collins
Gesendet: Dienstag, 16. Dezember 2008 15:37
An:  <mailto:freeswitch-users at lists.freeswitch.org>
freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

Just a hunch but try removing the spaces in this line:

 

loadzone=de

Zaptel can be quirky. 

-MC


Sent from my iPhone


On Dec 16, 2008, at 5:54 AM, "fidibus83" < <mailto:fidibus83 at aol.com>
fidibus83 at aol.com> wrote:

I have installed zaptel-1.4.11 

I have looked in zonedata.c and there is configured de-tonezone 

 


  _____  


Von:  <mailto:freeswitch-users-bounces at lists.freeswitch.org>
freeswitch-users-bounces at lists.freeswitch.org [
<mailto:freeswitch-users-bounces at lists.freeswitch.org>
mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael
Jerris
Gesendet: Dienstag, 16. Dezember 2008 14:20
An:  <mailto:freeswitch-users at lists.freeswitch.org>
<mailto:freeswitch-users at lists.freeswitch.org>
freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Zaptel Error!!!

 

It sounds like there is no de tonezone in the zaptel drivers, but I can't
imagine thats true.  What version of the drivers do you have installed?  

 

 

On Dec 16, 2008, at 4:54 AM, fidibus83 wrote:









Hello, I?m a newbie in FS and my English isn?t very good but I try to
explain my problem. Hopefully you can understand me. :-)

 

I have a Linux-Server with a Digium Wildcard TE110P. I install and configure
openzap (PRI/ISDN) and zaptel. But I have an error when I execute ztcfg ?vv:

 

31 channels configured.

 

ioctl(ZT_LOADZONE) failed: Invalid argument

Notice: Configuration file is /etc/zaptel.conf

line 288: Unable to register tone zone 'de'

 

 

 

ZAPTEL.CONF

 

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31

dchan=16

loadzone = de

defaultzone=de

 

What do I wrong? Please help me!

_______________________________________________
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<mailto:Freeswitch-users at lists.freeswitch.org>
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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
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http://lists.freeswitch.org/mailman/options/freeswitch-users
 <http://www.freeswitch.org>  <http://www.freeswitch.org>
http://www.freeswitch.org

 

= 

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 <http://www.freeswitch.org> http://www.freeswitch.org

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= y> = 

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From carole.olivier at enst.fr  Wed Dec 17 05:34:08 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 05:34:08 -0800 (PST)
Subject: [Freeswitch-users]  dynamic conference
Message-ID: <21053181.post@talk.nabble.com>


Hello,

I have done a small change in my dialplan which works but since I am new
with FreeSWITCH I was wondering if this solution goes with the philosophy of
the software or if it is absurd and there is a solution more adapted .

I try to reproduce the following functionality:
"A and B are on a simple call and decide to add C and have a conference.
Later on they decide also to invite D..."

In the dialplan I have added (based on the default dialplan):
<extension name="Local_Extension">
     <condition field="destination_number" expression="^(10[01][0-9])$">
           ................
          <anti-action application="bind_meta_app" data="1 a o transfer::
transf_call_${caller_id_number}"/>
          <anti-action application="bind_meta_app" data="2 a s
transfer::-both transf_call_${caller_id_number}"/>
          .......
      </condition>
</extension>

<extension name="transf_call_to_conf">
      <condition field="destination_number" expression="^transf_call_(.*)$">
         <action application="answer"/>
         <action application="conference" data="conf$1 at conf_dynamic"/>
       </condition>
</extension>

Here, if A calls B then A can bridge both legs into a conference named
"confnumberofA" by dialing *2.
Then, A can put this call on hold and call C. 
- If C answers and agrees then A can press *1 in order to bridge C into the
same conference "confnumberofA". A will come back into the conference which
is still on hold. 
- If C does not answer then A will still be able to come back into the
conference it puts on hold.

My main problem is the name of the conference. Since everybody should be
able to convert a simple call into a conference, the conference's name has
to be unique each time. I have chosen to make it dependent on the caller
number which is not perfect because then he is the only one which can bridge
the call and add member. However, I do not have any other idea, maybe I have
missed another possibility.

I would be glad to get any critics you have, it would help me to better
understand the fundamental concepts.

Thanks,
Carole
-- 
View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21053181.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From vkobashi at ydeasolutions.com.br  Wed Dec 17 05:45:03 2008
From: vkobashi at ydeasolutions.com.br (vinicius)
Date: Wed, 17 Dec 2008 11:45:03 -0200
Subject: [Freeswitch-users] LDAP Integration
In-Reply-To: <49485AD1.5070708@skopis.com>
References: <49417123.10709@ydeasolutions.com.br>	<49417538.9040203@ydeasolutions.com.br>	<200812120842.00808.hads@nice.net.nz>	<49418790.60001@ydeasolutions.com.br>	<87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com>	<494198F3.10806@ydeasolutions.com.br>	<4947A61F.6060806@ydeasolutions.com.br>
	<49485AD1.5070708@skopis.com>
Message-ID: <4949025F.9040008@ydeasolutions.com.br>

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From freeswitch-users at lists.rupa.com  Wed Dec 17 05:47:45 2008
From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists))
Date: Wed, 17 Dec 2008 07:47:45 -0600
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21053181.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com>
Message-ID: <49490301.8040701@lists.rupa.com>

On 12/17/2008 7:34 AM, Carole O. wrote:
> My main problem is the name of the conference. Since everybody should be
> able to convert a simple call into a conference, the conference's name has
> to be unique each time. I have chosen to make it dependent on the caller
> number which is not perfect because then he is the only one which can bridge
> the call and add member. However, I do not have any other idea, maybe I have
> missed another possibility.

If the conf name has to be unique, why not ensure that by making the
conference name based on the uuid of the a-leg?

> Thanks,
> Carole




From anthony.minessale at gmail.com  Wed Dec 17 06:19:31 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Dec 2008 08:19:31 -0600
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <21050436.post@talk.nabble.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
	<191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
	<21050314.post@talk.nabble.com> <21050436.post@talk.nabble.com>
Message-ID: <191c3a030812170619l27c9a7b3n55321efa1fa80f60@mail.gmail.com>

I said i unblocked the ones in mod_commands

mod_conference was it's own module.
I changed it to work in latest trunk as well.



On Wed, Dec 17, 2008 at 4:15 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> (I have just read the post again, I have written <action
> application="bind_meta_app" data="1 a s conference::conf1 lock"/> but I
> meant <action application="bind_meta_app" data="2 a s conference::conf1
> lock"/> please don't pay attention for that, I made the mistake when I have
> copied it in the post, not in the configuration.)
>
>
> Carole O. wrote:
> >
> > Thanks, this works fine.
> >
> > But I try to use some other API commands and something goes wrong: I
> would
> > like to be able to use the API commands for the conference like lock,
> > unlock, say, etc... from the dialplan.
> >
> > I try to add in my dialplan
> > <action application="bind_meta_app" data="1 a s conference::conf1 lock"/>
> > but it did not work, I presse F8 and I have got:
> >
> > 2008-12-17 10:47:58 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
> > sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes
> Processing
> > meta digit '2' [conference::conf1 lock]
> > 2008-12-17 10:47:58 [DEBUG] switch_core_session.c:611
> > switch_core_session_queue_private_event() Kill
> > sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes [BREAK]
> >
> > You told me that the API commands should not be blocked isn't it? So I do
> > not understand where I am wrong. Is "bind_meta_app" not supported in
> > conference?
> >
> > I have also tried to write directly:
> > <action application="conference" data="conf1 lock"/>
> > which also has no effect.
> >
> > Thanks for your help,
> > Carole
> >
> >
> > Anthony Minessale-2 wrote:
> >>
> >> it should work with
> >> <action application="bind_meta_app" data="1 ab s transfer::-both 3333"/>
> >>
> >> make sure you have debug log on (press f8) to see if there are any
> >> issues.
> >>
> >>
> >> On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr>
> >> wrote:
> >>
> >>>
> >>> Thanks for this answer.
> >>> Just a question so that everything is clear: there is no command to
> >>> bridge
> >>> directly a call and both legs into a conference? We have to transfer
> the
> >>> call to another extension and from there create the conference isn't?
> >>>
> >>> I used the command transfer but I still have a problem. I do the
> >>> following
> >>> and it did not work: in my dialplan I write:
> >>> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
> >>>
> >>> The extension 3333 works fine.
> >>>
> >>> However, nothing happens when I press *1.
> >>> Do you have an idea where am I wrong?
> >>>
> >>> (A subsidiary question: will both legs be transferred to the extension
> >>> 3333
> >>> or just the one which press *1? is there a way to transfer both
> >>> together?)
> >>>
> >>>
> >>> Thanks a lot,
> >>> Carole
> >>>
> >>>
> >>> Anthony Minessale-2 wrote:
> >>> >
> >>> > There is a transfer dial plan application also.
> >>> >
> >>> > There is not really any benefit in blocking the api commands from the
> >>> > dialplan
> >>> > apart from the potential for a blocking api call to delay the audio
> >>> stream
> >>> > which
> >>> > you can do at your own risk and use the sleep application with 0ms to
> >>> > flush
> >>> > the read buffers.
> >>> >
> >>> > So I guess i can lift that limitation in tree.
> >>> >
> >>> > see r10790
> >>> >
> >>> >
> >>> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
> >>> wrote:
> >>> >
> >>> >>
> >>> >> Hello,
> >>> >>
> >>> >> I have a general question about the API commands. Some of them are
> >>> not
> >>> >> available in the dialplan like uuid_transfer. I was wondering how to
> >>> call
> >>> >> an
> >>> >> API command without using the CLI. Especially I would be interested
> >>> in
> >>> >> knowing if there is any way to call them from a phone, I mean bind a
> >>> key
> >>> >> to
> >>> >> an API command.
> >>> >>
> >>> >> For instance I would like to transfer both members of a simple call
> >>> into
> >>> >> a
> >>> >> conference by dialing *1.
> >>> >> I have seen the transfer is possible from the CLI by doing:
> >>> >> api uuid_transfer <uuid> -both 3001
> >>> >> but I do not know how to do it else.
> >>> >>
> >>> >> If somebody could give me an insight about the topic it would be
> >>> great.
> >>> >> Thanks,
> >>> >> Carole
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >> --
> >>> >> View this message in context:
> >>> >>
> >>>
> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
> >>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>> >>
> >>> >>
> >>> >> _______________________________________________
> >>> >> Freeswitch-users mailing list
> >>> >> Freeswitch-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> UNSUBSCRIBE:
> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >>
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Anthony Minessale II
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> >
> >>> > AIM: anthm
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> <MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> ><
> >>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> <MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >
> >>> >
> >>> >
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> <PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >
> >>> >
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> <sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> ><
> >>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> <sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >
> >>> >
> >>> > iax:guest at conference.freeswitch.org/888
> >>> >
> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> <googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >
> >>> >
> >>> > pstn:213-799-1400
> >>> >
> >>> > _______________________________________________
> >>> > Freeswitch-users mailing list
> >>> > Freeswitch-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> --
> >>> View this message in context:
> >>>
> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>>
> >>>
> >>> _______________________________________________
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >> pstn:213-799-1400
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/general-question-about-API-command-tp21032754p21050436.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From carole.olivier at enst.fr  Wed Dec 17 06:24:28 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 06:24:28 -0800 (PST)
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <49490301.8040701@lists.rupa.com>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
Message-ID: <21054080.post@talk.nabble.com>


It would be unique you are right but I am not sure I can get its value if A
puts the call on hold, calls C and wants to add it to the conference whose
name dependent of the uuid of another session. 
I think if I use ${uuid} to add C I will have the uuid of the session
between A and C and not A and B no?
And I really have to configure this from the dialplan so statically.

Am I wrong somewhere??

Carole


Rupa Schomaker (lists)-2 wrote:
> 
> On 12/17/2008 7:34 AM, Carole O. wrote:
>> My main problem is the name of the conference. Since everybody should be
>> able to convert a simple call into a conference, the conference's name
>> has
>> to be unique each time. I have chosen to make it dependent on the caller
>> number which is not perfect because then he is the only one which can
>> bridge
>> the call and add member. However, I do not have any other idea, maybe I
>> have
>> missed another possibility.
> 
> If the conf name has to be unique, why not ensure that by making the
> conference name based on the uuid of the a-leg?
> 
>> Thanks,
>> Carole
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21054080.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From carole.olivier at enst.fr  Wed Dec 17 06:31:48 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Wed, 17 Dec 2008 06:31:48 -0800 (PST)
Subject: [Freeswitch-users] general question about API command
In-Reply-To: <191c3a030812170619l27c9a7b3n55321efa1fa80f60@mail.gmail.com>
References: <21032754.post@talk.nabble.com>
	<191c3a030812160548o11e42d94u26aafde7c5e3f7e7@mail.gmail.com>
	<21033993.post@talk.nabble.com>
	<191c3a030812160717k5017ca5bla4fbb9132267fec7@mail.gmail.com>
	<21050314.post@talk.nabble.com> <21050436.post@talk.nabble.com>
	<191c3a030812170619l27c9a7b3n55321efa1fa80f60@mail.gmail.com>
Message-ID: <21054214.post@talk.nabble.com>


ok 
Thanks a lot,
Carole


Anthony Minessale-2 wrote:
> 
> I said i unblocked the ones in mod_commands
> 
> mod_conference was it's own module.
> I changed it to work in latest trunk as well.
> 
> 
> 
> On Wed, Dec 17, 2008 at 4:15 AM, Carole O. <carole.olivier at enst.fr> wrote:
> 
>>
>> (I have just read the post again, I have written <action
>> application="bind_meta_app" data="1 a s conference::conf1 lock"/> but I
>> meant <action application="bind_meta_app" data="2 a s conference::conf1
>> lock"/> please don't pay attention for that, I made the mistake when I
>> have
>> copied it in the post, not in the configuration.)
>>
>>
>> Carole O. wrote:
>> >
>> > Thanks, this works fine.
>> >
>> > But I try to use some other API commands and something goes wrong: I
>> would
>> > like to be able to use the API commands for the conference like lock,
>> > unlock, say, etc... from the dialplan.
>> >
>> > I try to add in my dialplan
>> > <action application="bind_meta_app" data="1 a s conference::conf1
>> lock"/>
>> > but it did not work, I presse F8 and I have got:
>> >
>> > 2008-12-17 10:47:58 [DEBUG] switch_ivr_async.c:1425 meta_on_dtmf()
>> > sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes
>> Processing
>> > meta digit '2' [conference::conf1 lock]
>> > 2008-12-17 10:47:58 [DEBUG] switch_core_session.c:611
>> > switch_core_session_queue_private_event() Kill
>> > sofia/internal/1002 at 192.168.1.102:2054;line=n7ovvfr7;fs_nat=yes [BREAK]
>> >
>> > You told me that the API commands should not be blocked isn't it? So I
>> do
>> > not understand where I am wrong. Is "bind_meta_app" not supported in
>> > conference?
>> >
>> > I have also tried to write directly:
>> > <action application="conference" data="conf1 lock"/>
>> > which also has no effect.
>> >
>> > Thanks for your help,
>> > Carole
>> >
>> >
>> > Anthony Minessale-2 wrote:
>> >>
>> >> it should work with
>> >> <action application="bind_meta_app" data="1 ab s transfer::-both
>> 3333"/>
>> >>
>> >> make sure you have debug log on (press f8) to see if there are any
>> >> issues.
>> >>
>> >>
>> >> On Tue, Dec 16, 2008 at 8:43 AM, Carole O. <carole.olivier at enst.fr>
>> >> wrote:
>> >>
>> >>>
>> >>> Thanks for this answer.
>> >>> Just a question so that everything is clear: there is no command to
>> >>> bridge
>> >>> directly a call and both legs into a conference? We have to transfer
>> the
>> >>> call to another extension and from there create the conference isn't?
>> >>>
>> >>> I used the command transfer but I still have a problem. I do the
>> >>> following
>> >>> and it did not work: in my dialplan I write:
>> >>> <action application="bind_meta_app" data="1 ab s transfer::3333"/>
>> >>>
>> >>> The extension 3333 works fine.
>> >>>
>> >>> However, nothing happens when I press *1.
>> >>> Do you have an idea where am I wrong?
>> >>>
>> >>> (A subsidiary question: will both legs be transferred to the
>> extension
>> >>> 3333
>> >>> or just the one which press *1? is there a way to transfer both
>> >>> together?)
>> >>>
>> >>>
>> >>> Thanks a lot,
>> >>> Carole
>> >>>
>> >>>
>> >>> Anthony Minessale-2 wrote:
>> >>> >
>> >>> > There is a transfer dial plan application also.
>> >>> >
>> >>> > There is not really any benefit in blocking the api commands from
>> the
>> >>> > dialplan
>> >>> > apart from the potential for a blocking api call to delay the audio
>> >>> stream
>> >>> > which
>> >>> > you can do at your own risk and use the sleep application with 0ms
>> to
>> >>> > flush
>> >>> > the read buffers.
>> >>> >
>> >>> > So I guess i can lift that limitation in tree.
>> >>> >
>> >>> > see r10790
>> >>> >
>> >>> >
>> >>> > On Tue, Dec 16, 2008 at 7:07 AM, Carole O. <carole.olivier at enst.fr>
>> >>> wrote:
>> >>> >
>> >>> >>
>> >>> >> Hello,
>> >>> >>
>> >>> >> I have a general question about the API commands. Some of them are
>> >>> not
>> >>> >> available in the dialplan like uuid_transfer. I was wondering how
>> to
>> >>> call
>> >>> >> an
>> >>> >> API command without using the CLI. Especially I would be
>> interested
>> >>> in
>> >>> >> knowing if there is any way to call them from a phone, I mean bind
>> a
>> >>> key
>> >>> >> to
>> >>> >> an API command.
>> >>> >>
>> >>> >> For instance I would like to transfer both members of a simple
>> call
>> >>> into
>> >>> >> a
>> >>> >> conference by dialing *1.
>> >>> >> I have seen the transfer is possible from the CLI by doing:
>> >>> >> api uuid_transfer <uuid> -both 3001
>> >>> >> but I do not know how to do it else.
>> >>> >>
>> >>> >> If somebody could give me an insight about the topic it would be
>> >>> great.
>> >>> >> Thanks,
>> >>> >> Carole
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >>
>> >>> >> --
>> >>> >> View this message in context:
>> >>> >>
>> >>>
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21032754.html
>> >>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >>> >>
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> Freeswitch-users mailing list
>> >>> >> Freeswitch-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> UNSUBSCRIBE:
>> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >>
>> >>> >
>> >>> >
>> >>> >
>> >>> > --
>> >>> > Anthony Minessale II
>> >>> >
>> >>> > FreeSWITCH http://www.freeswitch.org/
>> >>> > ClueCon http://www.cluecon.com/
>> >>> >
>> >>> > AIM: anthm
>> >>> >
>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>>
>> <MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> ><
>> >>>
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> <MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
>> >
>> >>> >
>> >>> >
>> >>>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> >>>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> <PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
>> >
>> >>> >
>> >>> > IRC: irc.freenode.net #freeswitch
>> >>> >
>> >>> > FreeSWITCH Developer Conference
>> >>> >
>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>>
>> <sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> ><
>> >>>
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> <sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
>> >
>> >>> >
>> >>> > iax:guest at conference.freeswitch.org/888
>> >>> >
>> >>>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> >>>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> <googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
>> >
>> >>> >
>> >>> > pstn:213-799-1400
>> >>> >
>> >>> > _______________________________________________
>> >>> > Freeswitch-users mailing list
>> >>> > Freeswitch-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> --
>> >>> View this message in context:
>> >>>
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21033993.html
>> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> Freeswitch-users mailing list
>> >>> Freeswitch-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com><
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >
>> >>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org><
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> >> iax:guest at conference.freeswitch.org/888
>> >>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> >> pstn:213-799-1400
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/general-question-about-API-command-tp21032754p21050436.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p21054214.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From cstomi.levlist at gmail.com  Wed Dec 17 06:49:34 2008
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Wed, 17 Dec 2008 15:49:34 +0100
Subject: [Freeswitch-users] DNS RV faiover
Message-ID: <4949117E.5090202@gmail.com>

Hello,

We'd like to use DNS SRV for failover.

if we are bridge sofia/profile/whatever at domain.with.srv it works perfectly
but with gateway wich has this record in its proxy parameter it doesn't 
work.
Once we set up an A record too it works, so we assume dialing gateway 
doesn't use SRV records.
Is there any differences between the 2 method?

Thanks any help,
Tamas






From helmut.kuper at ewetel.de  Wed Dec 17 06:53:38 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 17 Dec 2008 15:53:38 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <4947EBB2.7030401@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>	<4947CBCB.8060204@ewetel.de>	<4947DAE1.1050706@ewetel.de>	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
	<4947EBB2.7030401@ewetel.de>
Message-ID: <49491272.9000103@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

I updated the jira bug. I did a Q931/Q921 trace. Currently there is no
direct hint, that FS is doing something wrong. NT side is just not
anwering the SETUP of FS, buuut, I was asked if FS is able to allow NT
side the channel management, so NT says what channel is to use instead
of FS.


Can FS do this? If so, how can I configure that?

Analyzing the trace throws a new bug: Sometimes our EWSD finds a
checksum errors in FS's RR messages.

more here: http://jira.freeswitch.org/browse/OPENZAP-40


If one of FS board wants the whole decoded trace just raise your hand.

regards
Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.9 (MingW32)

iEYEARECAAYFAklJEnIACgkQ4tZeNddg3dxIdwCbBbTw7a3bG/mnVvVmDSpbH5Bw
SU0AniHPV4qTIpuI8ENclxXyOn6pcueR
=Ey+M
-----END PGP SIGNATURE-----



From anthony.minessale at gmail.com  Wed Dec 17 07:05:46 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Dec 2008 09:05:46 -0600
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21054080.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
	<21054080.post@talk.nabble.com>
Message-ID: <191c3a030812170705n686bb20buc2bbd002c36e4a49@mail.gmail.com>

you could make up a uuid just for the conference name in the original call

<action application="export" data="my_conf_id=${create_uuid()}"/>

Now this channel and any other channel created by this channel will inherit
this var


On Wed, Dec 17, 2008 at 8:24 AM, Carole O. <carole.olivier at enst.fr> wrote:

>
> It would be unique you are right but I am not sure I can get its value if A
> puts the call on hold, calls C and wants to add it to the conference whose
> name dependent of the uuid of another session.
> I think if I use ${uuid} to add C I will have the uuid of the session
> between A and C and not A and B no?
> And I really have to configure this from the dialplan so statically.
>
> Am I wrong somewhere??
>
> Carole
>
>
> Rupa Schomaker (lists)-2 wrote:
> >
> > On 12/17/2008 7:34 AM, Carole O. wrote:
> >> My main problem is the name of the conference. Since everybody should be
> >> able to convert a simple call into a conference, the conference's name
> >> has
> >> to be unique each time. I have chosen to make it dependent on the caller
> >> number which is not perfect because then he is the only one which can
> >> bridge
> >> the call and add member. However, I do not have any other idea, maybe I
> >> have
> >> missed another possibility.
> >
> > If the conf name has to be unique, why not ensure that by making the
> > conference name based on the uuid of the a-leg?
> >
> >> Thanks,
> >> Carole
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/dynamic-conference-tp21053181p21054080.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From anthony.minessale at gmail.com  Wed Dec 17 07:10:49 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Dec 2008 09:10:49 -0600
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <49491272.9000103@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
	<4947EBB2.7030401@ewetel.de> <49491272.9000103@ewetel.de>
Message-ID: <191c3a030812170710u1f85fe97s7121cda8df02a50@mail.gmail.com>

try

<param name="suggest-channel" value="true"/>

in the <span> in openzap.conf.xml


On Wed, Dec 17, 2008 at 8:53 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi,
>
> I updated the jira bug. I did a Q931/Q921 trace. Currently there is no
> direct hint, that FS is doing something wrong. NT side is just not
> anwering the SETUP of FS, buuut, I was asked if FS is able to allow NT
> side the channel management, so NT says what channel is to use instead
> of FS.
>
>
> Can FS do this? If so, how can I configure that?
>
> Analyzing the trace throws a new bug: Sometimes our EWSD finds a
> checksum errors in FS's RR messages.
>
> more here: http://jira.freeswitch.org/browse/OPENZAP-40
>
>
> If one of FS board wants the whole decoded trace just raise your hand.
>
> regards
> Helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklJEnIACgkQ4tZeNddg3dxIdwCbBbTw7a3bG/mnVvVmDSpbH5Bw
> SU0AniHPV4qTIpuI8ENclxXyOn6pcueR
> =Ey+M
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From anthony.minessale at gmail.com  Wed Dec 17 07:13:20 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Dec 2008 09:13:20 -0600
Subject: [Freeswitch-users] DNS RV faiover
In-Reply-To: <4949117E.5090202@gmail.com>
References: <4949117E.5090202@gmail.com>
Message-ID: <191c3a030812170713n31330e75o94b8c2ded1010973@mail.gmail.com>

it might not.

try putting the value in register-proxy as well
sip:host.tld


On Wed, Dec 17, 2008 at 8:49 AM, Tamas Cseke <cstomi.levlist at gmail.com>wrote:

> Hello,
>
> We'd like to use DNS SRV for failover.
>
> if we are bridge sofia/profile/whatever at domain.with.srv it works perfectly
> but with gateway wich has this record in its proxy parameter it doesn't
> work.
> Once we set up an A record too it works, so we assume dialing gateway
> doesn't use SRV records.
> Is there any differences between the 2 method?
>
> Thanks any help,
> Tamas
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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From freeswitch-users at lists.rupa.com  Wed Dec 17 07:18:31 2008
From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists))
Date: Wed, 17 Dec 2008 09:18:31 -0600
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21054080.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
	<21054080.post@talk.nabble.com>
Message-ID: <49491847.3030403@lists.rupa.com>

On 12/17/2008 8:24 AM, Carole O. wrote:
> It would be unique you are right but I am not sure I can get its value if A
> puts the call on hold, calls C and wants to add it to the conference whose
> name dependent of the uuid of another session. 
> I think if I use ${uuid} to add C I will have the uuid of the session
> between A and C and not A and B no?
> And I really have to configure this from the dialplan so statically.
> 
> Am I wrong somewhere??
> 
> Carole

Ah, yeah.  uuid would not be the same when initiating a new call that
you then transfer to the conference call.  You need something that is
intrinsic to the endpoint.

I did a quick info dump to an originated call.  Depending on your
use-case (are these calls originating from registered handsets, trunked
from a sip provider, etc) you might want to rely on the variable
"sip_contact_uri" which is a combination of registered user name and ip
(and port if port isn't 5060).  This should be unique per endpoint.





From helmut.kuper at ewetel.de  Wed Dec 17 07:30:22 2008
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 17 Dec 2008 16:30:22 +0100
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <191c3a030812170710u1f85fe97s7121cda8df02a50@mail.gmail.com>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>	<4947CBCB.8060204@ewetel.de>
	<4947DAE1.1050706@ewetel.de>	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>	<4947EBB2.7030401@ewetel.de>
	<49491272.9000103@ewetel.de>
	<191c3a030812170710u1f85fe97s7121cda8df02a50@mail.gmail.com>
Message-ID: <49491B0E.8080505@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Anthony,

thx, but that doesn't work very good. Outgoing calls ring only once and
then this error rises in console:

2008-12-17 16:23:19 [DEBUG] Span:0 Q.931() Sending message to Layer4
(size: 103)
2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got
an event! Type:[02] Size:[103] CRV: 7 (0x7, CTX: Terminator)
2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan
810f590 (1:1) source isdn_data->channels_local_crv[0x7]
2008-12-17 16:23:19 [CRIT] ozmod_isdn.c:701 zap_isdn_931_34() Received
CALL PROCEEDING message for channel 0
2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:702 zap_isdn_931_34() Changing
state on 1:1 from DIALING to PROGRESS

regards
Helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.9 (MingW32)

iEYEARECAAYFAklJGw4ACgkQ4tZeNddg3dyeoACfeM6hYQF45T2gg18RQsOpZIjS
SB0AoIQp+ixmWUGBKyMFXIZQ6AbQsWK0
=I2L3
-----END PGP SIGNATURE-----



From cstomi.levlist at gmail.com  Wed Dec 17 08:05:37 2008
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Wed, 17 Dec 2008 17:05:37 +0100
Subject: [Freeswitch-users] DNS RV faiover
In-Reply-To: <191c3a030812170713n31330e75o94b8c2ded1010973@mail.gmail.com>
References: <4949117E.5090202@gmail.com>
	<191c3a030812170713n31330e75o94b8c2ded1010973@mail.gmail.com>
Message-ID: <49492351.4070101@gmail.com>

Helo,

with register-proxy registrations use SRV.
but it I use register=false param, and dial the gw it lookup only A.

I figured out if I don't specify the port in proxy param the SRV lookup 
is working,
but if I put ":5060" it doen't work. So it is a problem if I don't want 
to use the default port.

Regards,
Tamas

Anthony Minessale ?rta:
> it might not.
>
> try putting the value in register-proxy as well
> sip:host.tld
>
>
> On Wed, Dec 17, 2008 at 8:49 AM, Tamas Cseke <cstomi.levlist at gmail.com>wrote:
>
>   
>> Hello,
>>
>> We'd like to use DNS SRV for failover.
>>
>> if we are bridge sofia/profile/whatever at domain.with.srv it works perfectly
>> but with gateway wich has this record in its proxy parameter it doesn't
>> work.
>> Once we set up an A record too it works, so we assume dialing gateway
>> doesn't use SRV records.
>> Is there any differences between the 2 method?
>>
>> Thanks any help,
>> Tamas
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From brian at freeswitch.org  Wed Dec 17 08:12:22 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Dec 2008 10:12:22 -0600
Subject: [Freeswitch-users] DNS RV faiover
In-Reply-To: <49492351.4070101@gmail.com>
References: <4949117E.5090202@gmail.com>
	<191c3a030812170713n31330e75o94b8c2ded1010973@mail.gmail.com>
	<49492351.4070101@gmail.com>
Message-ID: <040BE086-EDF5-41C3-A805-C651CC7ED126@freeswitch.org>


On Dec 17, 2008, at 10:05 AM, Tamas Cseke wrote:

> Helo,
>
> with register-proxy registrations use SRV.
> but it I use register=false param, and dial the gw it lookup only A.
>
> I figured out if I don't specify the port in proxy param the SRV  
> lookup
> is working,
> but if I put ":5060" it doen't work. So it is a problem if I don't  
> want
> to use the default port.

Thats exactly what it should do you told it the port and address so  
its not going to lookup any SRV records.

/b



>
>
> Regards,
> Tamas




From kristjan.ugrin at gmail.com  Wed Dec 17 06:46:16 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Wed, 17 Dec 2008 15:46:16 +0100
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
Message-ID: <op.uma9zdmxatcbto@linux.site>

I have an idea which is takes too many characters for irc.
I'm relatively new to telephony and such stuff, I managed to get  
freeswitch running, but I don't fully uderstand
my problem in detail and how to solve it, so I need a bit of directions.

Briefly, my idea is to have a jabber contact, which would gtalk user add  
as their buddy. Let's call it callbuddy at somejabbersrv.com.
This would be a bot that would accept commands - I've already made a small  
xmpp java bot which just logs into gtalk and send echo
messages to users - like a parrot, nothing serious. Also I've set up  
openfire jabber server.
What I would like him to do, is when user would tipe "call 0189432443" it  
would initiate a call between contact who tiped in this number (command)  
and this number.

Is this possible, and what would be the best implementation (dingaling  
acting as client...)?

Thanks.

-- 
kriko



From freeswitch at davidnicol.otherinbox.com  Wed Dec 17 07:41:46 2008
From: freeswitch at davidnicol.otherinbox.com (freeswitch at davidnicol.otherinbox.com)
Date: Wed, 17 Dec 2008 10:41:46 -0500
Subject: [Freeswitch-users] Cisco contest
Message-ID: <200812171541.mBHFfj4Y010761@box7.911domain.com>

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From mike at jerris.com  Wed Dec 17 09:10:18 2008
From: mike at jerris.com (Michael Jerris)
Date: Wed, 17 Dec 2008 12:10:18 -0500
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <20081217074002.GA16365@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
Message-ID: <5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>


On Dec 17, 2008, at 2:40 AM, Jason White wrote:

> The code in bind6only_check in libs/sofia-sip/libsofia-sip-ua/tport/ 
> tport.c looks
> correct to me, but I can't find where the result is tested (it's in
> mr_bindv6only). When bind6only_check() is called in  
> tport_bind_server(), the
> return value isn't tested, and I'm having difficulty finding where  
> it is used
> - I'm interested in whether we're in fact attempting to bind only to  
> the IPv6
>  port and whether the logic is correct here.
>
> When I find the time, I could rebuild with debug symbols and run it  
> under gdb.

If this is in fact a bug, could you please report it to the sofia-sip  
bugtracker.  Patches are very helpful there.

Mike




From mike at jerris.com  Wed Dec 17 09:12:42 2008
From: mike at jerris.com (Michael Jerris)
Date: Wed, 17 Dec 2008 12:12:42 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
In-Reply-To: <8CB2E358C58A406-9B8-1883@WEBMAIL-MA13.sysops.aol.com>
References: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com>
	<957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com>
	<8CB2E358C58A406-9B8-1883@WEBMAIL-MA13.sysops.aol.com>
Message-ID: <674E25FB-0584-4357-B556-81BF79F84209@jerris.com>

I think the best way to confirm all this is to load a full pcap in  
wireshark and have it pull the wav file of the individual audio  
streams to see what is going on.

Mike

On Dec 17, 2008, at 3:06 AM, mszlazak at aol.com wrote:

> Hi Mike,
>
> That does get the audio go between the softphone and the application  
> (Voxeo's Prophecy ASR) "around" FreeSwitch but I would like the  
> audio going "through" FreeSwitch. I plan to do something to it  
> before passing it on.
>
> Support from Voxeo had this to say about the "bypass media" setting  
> and if you could add some more insight that would be much  
> appreciated. Since this is all on one Windows XP machine they can't  
> get the info from the pcap file and are requesting I set up  
> freeswitch on another machine which I will do. I thought you may  
> have some more input.

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From mike at jerris.com  Wed Dec 17 09:15:20 2008
From: mike at jerris.com (Michael Jerris)
Date: Wed, 17 Dec 2008 12:15:20 -0500
Subject: [Freeswitch-users] Zaptel Error!!!
In-Reply-To: <004201c96030$b2b9fd80$6445310a@Franzi>
References: <002101c95f64$56958e60$6445310a@Franzi><7E59954F-69F7-4631-BEE3-F288B67BB3E4@jerris.com><007601c95f85$c1462aa0$6445310a@Franzi><9C7C17E7-DB40-4159-B878-8A91F2BCD2A4@freeswitch.org><009f01c95f8d$7918ddb0$6445310a@Franzi>
	<B4F10400-D789-4507-8C06-F2B7F79ECED7@freeswitch.org>
	<004201c96030$b2b9fd80$6445310a@Franzi>
Message-ID: <C3A43C47-73FA-4F63-B8D6-1BB053B9207F@jerris.com>

FreeSWITCH/openzap is completely uninvolved at this point, you might  
try asking on the zaptel mailing lists?

Mike

On Dec 17, 2008, at 5:17 AM, fidibus83 wrote:

> I did a reinstall but there is the same error!
> Is there something else I can do to remove the error?
>
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From kirk.bateman at gmail.com  Wed Dec 17 09:15:11 2008
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Wed, 17 Dec 2008 17:15:11 +0000
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
Message-ID: <2bee4fc40812170915p1b5d91feu5fcfbee6713fad40@mail.gmail.com>

Kriko,

I have been looking at the same sort of thing, but I'm planning to implement
an ejabberd bot component (so I can hopefully use the new mod_erlang_event
freeswitch interface).

It seems to me that bits of the current dingaling / jingle interface are
having problems, like not liking sending messages to other domains, its
generally working if they are all .gmail.com users but when you have some on
googlemail.com etc it starts breaking, and doesn't use the whole JID in the
from attribute for sending messages.

When I get a chance I'll try and narrow down the problem.

Cheers

Kirk

Date: Wed, 17 Dec 2008 15:46:16 +0100
> From: kriko <kristjan.ugrin at gmail.com>
> Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
> To: "freeswitch-users at lists.freeswitch.org"
>        <freeswitch-users at lists.freeswitch.org>
> Message-ID: <op.uma9zdmxatcbto at linux.site>
> Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8
>
> I have an idea which is takes too many characters for irc.
> I'm relatively new to telephony and such stuff, I managed to get
> freeswitch running, but I don't fully uderstand
> my problem in detail and how to solve it, so I need a bit of directions.
>
> Briefly, my idea is to have a jabber contact, which would gtalk user add
> as their buddy. Let's call it callbuddy at somejabbersrv.com.
> This would be a bot that would accept commands - I've already made a small
> xmpp java bot which just logs into gtalk and send echo
> messages to users - like a parrot, nothing serious. Also I've set up
> openfire jabber server.
> What I would like him to do, is when user would tipe "call 0189432443" it
> would initiate a call between contact who tiped in this number (command)
> and this number.
>
> Is this possible, and what would be the best implementation (dingaling
> acting as client...)?
>
> Thanks.
>
> --
> kriko
>
>
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From brian at freeswitch.org  Wed Dec 17 09:21:32 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Dec 2008 11:21:32 -0600
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
In-Reply-To: <2bee4fc40812170915p1b5d91feu5fcfbee6713fad40@mail.gmail.com>
References: <2bee4fc40812170915p1b5d91feu5fcfbee6713fad40@mail.gmail.com>
Message-ID: <876638A9-FBB0-4079-A76F-814E6953D395@freeswitch.org>

FreeSWITCH already logs into your jabber server as a component if you  
cant communicate with other domains then your jabber server is not  
configured correctly.

/b

On Dec 17, 2008, at 11:15 AM, Kirk Bateman wrote:

> Kriko,
>
> I have been looking at the same sort of thing, but I'm planning to  
> implement an ejabberd bot component (so I can hopefully use the new  
> mod_erlang_event freeswitch interface).
>
> It seems to me that bits of the current dingaling / jingle interface  
> are having problems, like not liking sending messages to other  
> domains, its generally working if they are all .gmail.com users but  
> when you have some ongooglemail.com etc it starts breaking, and  
> doesn't use the whole JID in the from attribute for sending messages.
>
> When I get a chance I'll try and narrow down the problem.
>
> Cheers
>
> Kirk

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From msc at freeswitch.org  Wed Dec 17 09:33:19 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 17 Dec 2008 09:33:19 -0800
Subject: [Freeswitch-users] OpenZap Not working Stable
In-Reply-To: <49491B0E.8080505@ewetel.de>
References: <e61df640809220729i4ab82409o38b74448b336d085@mail.gmail.com>
	<191c3a030809220836pdc290c5iaf7fdff1728b92d8@mail.gmail.com>
	<e61df640809230701k3f36f09fuf6905dd063eb9d01@mail.gmail.com>
	<191c3a030809230713r24c17e8bq7f948e0a4ca40b35@mail.gmail.com>
	<4947CBCB.8060204@ewetel.de> <4947DAE1.1050706@ewetel.de>
	<87f2f3b90812160900j7df3e624h1033745c6a795f4a@mail.gmail.com>
	<191c3a030812160935h72d0934fh11e58c0d6185f614@mail.gmail.com>
	<4947EBB2.7030401@ewetel.de> <49491272.9000103@ewetel.de>
	<191c3a030812170710u1f85fe97s7121cda8df02a50@mail.gmail.com>
	<49491B0E.8080505@ewetel.de>
Message-ID: <9C1B9E67-82A8-4FA0-BE62-01836FDC5594@freeswitch.org>

Helmut,

Can you turn on full debug and capture the output? It's a lot so put  
it in a pastebin.

-MC

Sent from my iPhone

On Dec 17, 2008, at 7:30 AM, Helmut Kuper <helmut.kuper at ewetel.de>  
wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Anthony,
>
> thx, but that doesn't work very good. Outgoing calls ring only once  
> and
> then this error rises in console:
>
> 2008-12-17 16:23:19 [DEBUG] Span:0 Q.931() Sending message to Layer4
> (size: 103)
> 2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I  
> got
> an event! Type:[02] Size:[103] CRV: 7 (0x7, CTX: Terminator)
> 2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan
> 810f590 (1:1) source isdn_data->channels_local_crv[0x7]
> 2008-12-17 16:23:19 [CRIT] ozmod_isdn.c:701 zap_isdn_931_34() Received
> CALL PROCEEDING message for channel 0
> 2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:702 zap_isdn_931_34()  
> Changing
> state on 1:1 from DIALING to PROGRESS
>
> regards
> Helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklJGw4ACgkQ4tZeNddg3dyeoACfeM6hYQF45T2gg18RQsOpZIjS
> SB0AoIQp+ixmWUGBKyMFXIZQ6AbQsWK0
> =I2L3
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From stephen at stephenjc.com  Wed Dec 17 10:00:09 2008
From: stephen at stephenjc.com (stephen at stephenjc)
Date: Wed, 17 Dec 2008 13:00:09 -0500
Subject: [Freeswitch-users] ignore dtmf
Message-ID: <1d9d102c0812171000i53a9e7d0u9557099f2b79cfe@mail.gmail.com>

I have a click to click system written in javascript, so i call out
both legs of the call then bridge them. I am looking for a way to
ignore dtmf tones on 1 leg of the call.


Thanks,
Stephen C
-All of my email addresses go to the same place
-Save Paper, think before you print.



From brian at freeswitch.org  Wed Dec 17 10:32:00 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Dec 2008 12:32:00 -0600
Subject: [Freeswitch-users] Cisco contest
In-Reply-To: <200812171541.mBHFfj4Y010761@box7.911domain.com>
References: <200812171541.mBHFfj4Y010761@box7.911domain.com>
Message-ID: <782B82A4-FF05-4F56-AB10-E8D6C6BD8A7C@freeswitch.org>

Seems pretty sleazy to me... what are they going to commercialize the  
results?

/b

On Dec 17, 2008, at 9:41 AM, freeswitch at davidnicol.otherinbox.com wrote:

> http://www.google.com/search?q=cisco+linux+contest
>
>
> Although cisco already does VOIP stuff so they might have trouble  
> awarding prizes to a technology which would compete with themselves,  
> but what are they expecting, putting Linux boards in Cisco backplanes?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mszlazak at aol.com  Wed Dec 17 11:29:36 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 17 Dec 2008 14:29:36 -0500
Subject: [Freeswitch-users] Help with routing sound locally through FS
In-Reply-To: <674E25FB-0584-4357-B556-81BF79F84209@jerris.com>
References: <8CB2D7C87FD57FD-388-EE@mblk-d47.sysops.aol.com><957547DE-C6CF-4C17-8718-6EFED2DECCA6@jerris.com><8CB2E358C58A406-9B8-1883@WEBMAIL-MA13.sysops.aol.com>
	<674E25FB-0584-4357-B556-81BF79F84209@jerris.com>
Message-ID: <8CB2E94EE5B5E49-E20-3F7@FWM-M37.sysops.aol.com>


 


 Hi Mike,

A brief talk with one of the Prophecy support people makes him think it maybe more a FreeSwitch issue but he wasn't that's been following my problem.

I've attached a pcap and FS log file from a past session. Also, there is a file with netstat's output.

The impression is that FS is not forwarding the audio to Prophecy correctly so Prophecy is timing out in different parts of it's dialogue before the hang up.

On the other hand, there is no problem when using "bypass media" but of course I can't use this if the media is being processed by FS before it's past on to.

Could please take a look.

Thanks. Mark. 


 

-----Original Message-----
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Wed, 17 Dec 2008 9:12 am
Subject: Re: [Freeswitch-users] Help with routing sound locally through FS









I think the best way to confirm all this is to load a full pcap in wireshark and have it pull the wav file of the individual audio streams to see what is going on.



Mike




On Dec 17, 2008, at 3:06 AM, mszlazak at aol.com wrote:



 Hi Mike,
 
 That does get the audio go between the softphone and the application (Voxeo's Prophecy ASR) "around" FreeSwitch but I would like the audio going "through" FreeSwitch. I plan to do something to it before passing it on.
 
 Support from Voxeo had this to say about the "bypass media" setting and if you could add some more insight that would be much appreciated. Since this is all on one Windows XP machine they can't get the info from the pcap file and are requesting I set up freeswitch on another machine which I will do. I thought you may have some more input.





=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 
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From itsc99 at cantv.net  Wed Dec 17 11:36:56 2008
From: itsc99 at cantv.net (Tjapko Smits)
Date: Wed, 17 Dec 2008 15:06:56 -0430
Subject: [Freeswitch-users] multi domain issue
Message-ID: <1229542616.24489.30.camel@tjaracas-main>

Hi, 

Fresh freeswitch user. Installed freeswitch with default installation a
week ago. Need more information on multi domain usage. Followed the wiki
pages with multi tenant examples. All working well but I do have a
problem when calling to an internal extension. 

Scenario: 

domain_A created by copying the default.xml to domain_A.xml following
steps from the wiki multi tenant informaion. Than created the domain_A
directory and copied the /default directory extensions there. 
Same steps for Domain_B. For the rest all is like default. 

Registered 3 phones to Domain_A and 3 phones to Domain_B 

Domain_A -> 1000 1005 and 1006 

Domain_B -> 1002 1005 and 1006

When I call from 1002 inside Domain_B to extension 1000 in Domain_A
phone rings and this is what I do not like to happen. 

Trace shows that when called to 1000 or 1005 or 1006 only the IP
addresses from those endpoints in Domain_A are addressed.

The first INVITE shows an OK invite with correct domain name but after
the 100 trying the re-invite makes an IP address out of most certainly
because it followed the rules in the dialplan.   

Can anybody point me in the direction how to make it possible that calls
from DOmain_A stay in Domain_A- and Domain_B stay in B etc. 

All the rest like inbound and outbound calls on dedicated gateways are
working fine. 

I assume that I need to configure acl for this but still not very clear.
Any tip will be most appreciated.  

-- 
Tjapko





From oseslija at gmail.com  Wed Dec 17 13:38:33 2008
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 17 Dec 2008 22:38:33 +0100
Subject: [Freeswitch-users] multi domain issue
In-Reply-To: <1229542616.24489.30.camel@tjaracas-main>
References: <1229542616.24489.30.camel@tjaracas-main>
Message-ID: <4468a6770812171338m153667ebnc567716e47dbc44f@mail.gmail.com>

Hi,

I have multi domain, multi tenant setup configured and working.

Did you add something like

    <param name="alias" value="sip:domain_A:5060"/>
    <param name="alias" value="sip:domain_B:5060"/>

to one of the profile configs for multi-domain so FreeSWITCH can look its
configs for those domains?
Also, check if you specified domain_A for "domain_name" param in the
domain_A.xml file.
Directory extensions context for other domains should not use the same
"user_context" param for they will hit default dialplan as well.

Please join the IRC channel (#freeswitch) for further questions you might
have.

Regards,
Ognjen
(sekil)
On Wed, Dec 17, 2008 at 8:36 PM, Tjapko Smits <itsc99 at cantv.net> wrote:

> Hi,
>
> Fresh freeswitch user. Installed freeswitch with default installation a
> week ago. Need more information on multi domain usage. Followed the wiki
> pages with multi tenant examples. All working well but I do have a
> problem when calling to an internal extension.
>
> Scenario:
>
> domain_A created by copying the default.xml to domain_A.xml following
> steps from the wiki multi tenant informaion. Than created the domain_A
> directory and copied the /default directory extensions there.
> Same steps for Domain_B. For the rest all is like default.
>
> Registered 3 phones to Domain_A and 3 phones to Domain_B
>
> Domain_A -> 1000 1005 and 1006
>
> Domain_B -> 1002 1005 and 1006
>
> When I call from 1002 inside Domain_B to extension 1000 in Domain_A
> phone rings and this is what I do not like to happen.
>
> Trace shows that when called to 1000 or 1005 or 1006 only the IP
> addresses from those endpoints in Domain_A are addressed.
>
> The first INVITE shows an OK invite with correct domain name but after
> the 100 trying the re-invite makes an IP address out of most certainly
> because it followed the rules in the dialplan.
>
> Can anybody point me in the direction how to make it possible that calls
> from DOmain_A stay in Domain_A- and Domain_B stay in B etc.
>
> All the rest like inbound and outbound calls on dedicated gateways are
> working fine.
>
> I assume that I need to configure acl for this but still not very clear.
> Any tip will be most appreciated.
>
> --
> Tjapko
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Dec 17 18:08:17 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 17 Dec 2008 18:08:17 -0800
Subject: [Freeswitch-users] ignore dtmf
In-Reply-To: <1d9d102c0812171000i53a9e7d0u9557099f2b79cfe@mail.gmail.com>
References: <1d9d102c0812171000i53a9e7d0u9557099f2b79cfe@mail.gmail.com>
Message-ID: <459F63E0-9E26-41DA-B5DD-FB95ACFC132C@freeswitch.org>

By ignore do you mean filter out? Or do you mean don't do anything but  
do audiblize the tones? Do you have some sort of application that does  
something with dtmfs?

-MC

Sent from my iPhone

On Dec 17, 2008, at 10:00 AM, "stephen at stephenjc" <stephen at stephenjc.com 
 > wrote:

> I have a click to click system written in javascript, so i call out
> both legs of the call then bridge them. I am looking for a way to
> ignore dtmf tones on 1 leg of the call.
>
>
> Thanks,
> Stephen C
> -All of my email addresses go to the same place
> -Save Paper, think before you print.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From stephen at stephenjc.com  Wed Dec 17 18:39:31 2008
From: stephen at stephenjc.com (stephen at stephenjc)
Date: Wed, 17 Dec 2008 21:39:31 -0500
Subject: [Freeswitch-users] ignore dtmf
In-Reply-To: <459F63E0-9E26-41DA-B5DD-FB95ACFC132C@freeswitch.org>
References: <1d9d102c0812171000i53a9e7d0u9557099f2b79cfe@mail.gmail.com>
	<459F63E0-9E26-41DA-B5DD-FB95ACFC132C@freeswitch.org>
Message-ID: <1d9d102c0812171839n3102ec33wf575fab7a9a73cfa@mail.gmail.com>

I have a click2click app that originates both legs of the call. After
the call is bridged if you press the # key it records the call. Only
the original person should be able to click # to enable records. That
is why i want to filter dtmf on 1 leg of the call.

Thanks,
Stephen C
-All of my email addresses go to the same place
-Save Paper, think before you print.




On Wed, Dec 17, 2008 at 9:08 PM, Michael S Collins <msc at freeswitch.org> wrote:
> By ignore do you mean filter out? Or do you mean don't do anything but
> do audiblize the tones? Do you have some sort of application that does
> something with dtmfs?
>
> -MC
>
> Sent from my iPhone
>
> On Dec 17, 2008, at 10:00 AM, "stephen at stephenjc" <stephen at stephenjc.com
>  > wrote:
>
>> I have a click to click system written in javascript, so i call out
>> both legs of the call then bridge them. I am looking for a way to
>> ignore dtmf tones on 1 leg of the call.
>>
>>
>> Thanks,
>> Stephen C
>> -All of my email addresses go to the same place
>> -Save Paper, think before you print.
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From scott.ellis at novatex.com.au  Wed Dec 17 18:45:26 2008
From: scott.ellis at novatex.com.au (Scott Ellis)
Date: Thu, 18 Dec 2008 13:45:26 +1100
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
In-Reply-To: <153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>
References: <49478B0F.3000802@novatex.com.au>
	<153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>
Message-ID: <4949B946.5050502@novatex.com.au>

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From markmorreny at gmail.com  Thu Dec 18 00:01:07 2008
From: markmorreny at gmail.com (mark morreny)
Date: Thu, 18 Dec 2008 16:01:07 +0800
Subject: [Freeswitch-users] Question about running Freeswitch in the
	background
Message-ID: <20ad6b920812180001t2ef2a2c7occ8130c57c69f3f5@mail.gmail.com>

Hi,

I have a small question about running Freeswich in the background using -nc
option.  Does freeswitch writes the log in any log file when running in
background mode?
If I have some print out statement in my custom mod, can I still see the
output of those logs somewhere?

Thanks,
Mark
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From jason at jasonjgw.net  Thu Dec 18 00:29:07 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 18 Dec 2008 19:29:07 +1100
Subject: [Freeswitch-users] Question about running Freeswitch in
	the	background
In-Reply-To: <20ad6b920812180001t2ef2a2c7occ8130c57c69f3f5@mail.gmail.com>
References: <20ad6b920812180001t2ef2a2c7occ8130c57c69f3f5@mail.gmail.com>
Message-ID: <20081218082907.GA11459@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 04:01:07PM +0800, mark morreny wrote:
> I have a small question about running Freeswich in the background using -nc
> option.  Does freeswitch writes the log in any log file when running in
> background mode?

Yes. Have a look at ~freeswitch/log/freeswitch.log





From kristjan.ugrin at gmail.com  Thu Dec 18 00:37:38 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 18 Dec 2008 09:37:38 +0100
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
In-Reply-To: <876638A9-FBB0-4079-A76F-814E6953D395@freeswitch.org>
References: <2bee4fc40812170915p1b5d91feu5fcfbee6713fad40@mail.gmail.com>
	<876638A9-FBB0-4079-A76F-814E6953D395@freeswitch.org>
Message-ID: <op.umcnkzr2atcbto@linux.site>

I'm not sure if we understood each other correctly.
I meant calling from jabber to other sip phones. I'm not sure you can just  
add a sip phone as
a buddy into e.g. gtalk or any other jabber service and call it.
So that why a bot (and only one but), which you would have as a buddy and  
you would feed him with numbers.
And he would estabilish a call between jabber user and typed in number.
Was I clearer?

I'll take a look how the component mode works, hopefully there is more  
documentation than only this:
http://wiki.freeswitch.org/wiki/Dingaling

Also if this functionality is not possible with fs as it is, I could maybe  
write a java program which interact with fs
interface and do that as intended?

Cheers

On Wed, 17 Dec 2008 18:21:32 +0100, Brian West <brian at freeswitch.org>  
wrote:

> FreeSWITCH already logs into your jabber server as a component if you
> cant communicate with other domains then your jabber server is not
> configured correctly.
>
> /b
>
> On Dec 17, 2008, at 11:15 AM, Kirk Bateman wrote:
>
>> Kriko,
>>
>> I have been looking at the same sort of thing, but I'm planning to
>> implement an ejabberd bot component (so I can hopefully use the new
>> mod_erlang_event freeswitch interface).
>>
>> It seems to me that bits of the current dingaling / jingle interface
>> are having problems, like not liking sending messages to other
>> domains, its generally working if they are all .gmail.com users but
>> when you have some ongooglemail.com etc it starts breaking, and
>> doesn't use the whole JID in the from attribute for sending messages.
>>
>> When I get a chance I'll try and narrow down the problem.
>>
>> Cheers
>>
>> Kirk
>



-- 
kriko



From jason at jasonjgw.net  Thu Dec 18 00:53:23 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 18 Dec 2008 19:53:23 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
	<5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
Message-ID: <20081218085323.GA11635@jdc.jasonjgw.net>

On Wed, Dec 17, 2008 at 12:10:18PM -0500, Michael Jerris wrote:
> 
> On Dec 17, 2008, at 2:40 AM, Jason White wrote:
> 
> > The code in bind6only_check in libs/sofia-sip/libsofia-sip-ua/tport/ 
> > tport.c looks
> > correct to me, but I can't find where the result is tested (it's in
> > mr_bindv6only). When bind6only_check() is called in  
> > tport_bind_server(), the
> > return value isn't tested, and I'm having difficulty finding where  
> > it is used

I realized after posting that if the IPv4 port is bound by another process,
then the attempt to bind to the IPv4 port in bind6only_check() should return
-1, and hence the result of bind6only_check() will be 0, even if the os allows
the IPv6 and IPv4 ports to be bound independently of each other.

This looks like a potential bug, but I haven't investigated properly to be
sure, and I'm extremely busy just now (as well as not being very experienced
at this).

> If this is in fact a bug, could you please report it to the sofia-sip  
> bugtracker.  Patches are very helpful there.

If anyone else has a chance to look at it before I do, please let me know.
Otherwise, I'll check it out when I have time to build a version of FreeSWITCH
with debug symbols and run it under gdb.




From brian at freeswitch.org  Thu Dec 18 01:01:12 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 03:01:12 -0600
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <20081218085323.GA11635@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
	<5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
	<20081218085323.GA11635@jdc.jasonjgw.net>
Message-ID: <0B5DCBAA-5FA5-488A-8189-39878A5FCDA0@freeswitch.org>

I bind mine independently without a problem on CentOS 5.2

/b

On Dec 18, 2008, at 2:53 AM, Jason White wrote:

> I realized after posting that if the IPv4 port is bound by another  
> process,
> then the attempt to bind to the IPv4 port in bind6only_check()  
> should return
> -1, and hence the result of bind6only_check() will be 0, even if the  
> os allows
> the IPv6 and IPv4 ports to be bound independently of each other.




From jason at jasonjgw.net  Thu Dec 18 01:11:49 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 18 Dec 2008 20:11:49 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <0B5DCBAA-5FA5-488A-8189-39878A5FCDA0@freeswitch.org>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
	<5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
	<20081218085323.GA11635@jdc.jasonjgw.net>
	<0B5DCBAA-5FA5-488A-8189-39878A5FCDA0@freeswitch.org>
Message-ID: <20081218091149.GA11826@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 03:01:12AM -0600, Brian West wrote:
> I bind mine independently without a problem on CentOS 5.2

Thanks; friends of mine have access to Fedora boxes, so we'll compare
behaviour and try to sort it out.




From fidibus83 at aol.com  Thu Dec 18 01:20:24 2008
From: fidibus83 at aol.com (fidibus83)
Date: Thu, 18 Dec 2008 10:20:24 +0100
Subject: [Freeswitch-users] SQL Error
Message-ID: <004801c960f1$dc680840$6445310a@Franzi>

Hello,

 

I?m a newbie in FS. I get an error from the freeswitch cli:

 

2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL ERR
[database disk image is malformed]

 

I don?t know what to do to remove this error! Can you help me?

 

Regards
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From hads at nice.net.nz  Thu Dec 18 01:37:33 2008
From: hads at nice.net.nz (Hadley Rich)
Date: Thu, 18 Dec 2008 22:37:33 +1300
Subject: [Freeswitch-users] SQL Error
In-Reply-To: <004801c960f1$dc680840$6445310a@Franzi>
References: <004801c960f1$dc680840$6445310a@Franzi>
Message-ID: <200812182237.33739.hads@nice.net.nz>

On Thursday 18 December 2008 22:20:24 fidibus83 wrote:
> I?m a newbie in FS. I get an error from the freeswitch cli:
>
> 2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL
> ERR [database disk image is malformed]
>
> I don?t know what to do to remove this error! Can you help me?

If you remove the database files (which are in $PREFIX/db) then FreeSWITCH will 
recreate then at startup.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier



From carole.olivier at enst.fr  Thu Dec 18 01:37:42 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Thu, 18 Dec 2008 01:37:42 -0800 (PST)
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <49491847.3030403@lists.rupa.com>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
	<21054080.post@talk.nabble.com> <49491847.3030403@lists.rupa.com>
Message-ID: <21069519.post@talk.nabble.com>


Hello,

Thanks for your answers!
Concerning the creation of a new variable for the conference the problem is
that I do not create channels from the conference. I call separately a new
member on a new channel and add it on the conference only if he agrees to
enter it. So it was the same problem as for the uuid, I am not sure I can
access the good variable from anywhere in case many conferences are running.

So, I do the following if somebody is interested in:

<extension name="Local_Extension">
     .....................
      <condition field="destination_number"
expression="^${caller_id_number}$">
        <anti-action application="bind_meta_app" data="1 a o transfer::
transf_opp_leg_${db(select/conference/${caller_id_number})}_${destination_number}"/>
        <anti-action application="bind_meta_app" data="2 ab s
transfer::-both
transf_both_${uuid}_${caller_id_number}_${destination_number}"/>
       ..........
</extension>

<extension name="transf_both_legs">
     <condition field="destination_number"
expression="^transf_both_(.*)_(.*)_(.*)$">
         <action application="db" data="insert/conference/$2/conf_$1"/>
         <action application="db" data="insert/conference/$3/conf_$1"/>
         <action application="answer"/>
         <action application="conference" data="conf_$1 at conf_dynamic"/>
     </condition>
</extension>

<extension name="transf_opposite_leg">
     <condition field="destination_number"
expression="^transf_opp_leg_(.*)_(.*)$">
         <action application="db" data="insert/conference/$2/$1"/>
         <action application="answer"/>
         <action application="conference" data="$1 at conf_dynamic"/>
     </condition>
</extension>

This is not perfect and I am not sure it is "cleanly" programmed but it
works and it is flexible. All the members in the conference can invite new
members thanks to the conference name they all have in the database.

If you still have critics they are all welcome!
Thanks for your help,
Best regards,
Carole


Rupa Schomaker (lists)-2 wrote:
> 
> On 12/17/2008 8:24 AM, Carole O. wrote:
>> It would be unique you are right but I am not sure I can get its value if
>> A
>> puts the call on hold, calls C and wants to add it to the conference
>> whose
>> name dependent of the uuid of another session. 
>> I think if I use ${uuid} to add C I will have the uuid of the session
>> between A and C and not A and B no?
>> And I really have to configure this from the dialplan so statically.
>> 
>> Am I wrong somewhere??
>> 
>> Carole
> 
> Ah, yeah.  uuid would not be the same when initiating a new call that
> you then transfer to the conference call.  You need something that is
> intrinsic to the endpoint.
> 
> I did a quick info dump to an originated call.  Depending on your
> use-case (are these calls originating from registered handsets, trunked
> from a sip provider, etc) you might want to rely on the variable
> "sip_contact_uri" which is a combination of registered user name and ip
> (and port if port isn't 5060).  This should be unique per endpoint.
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21069519.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From kirk.bateman at gmail.com  Thu Dec 18 01:45:54 2008
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Thu, 18 Dec 2008 09:45:54 +0000
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
Message-ID: <2bee4fc40812180145mda36e7jf1b2905b9a5c63af@mail.gmail.com>

Brian,

That wasn't exactly what I meant :)

I have had Freeswitch connecting to GTalk directly as a client and that was
where I was getting the issues with sending anything to other domains.

I haven't actually tried the server profile with my own ejabberd server.
What I was planning to do was have a separate ejabberd bot which is the
control interface which then sends bits to the freeswitch connected "bot" /
user so I can sanitise anything before it gets to freeswitch, as I have
found the freeswitch jabber stuff to segfault every now and then (that said
I haven't updated in about a month or so). Just haven't had enough time to
actually look into what was causing the segfaults.

I did get the gtalk to sip bits working though, had a few codec issues but
gtalk to freeswitch worked fine, unfortunately my slightly more complicated
gtalk to freeswitch to asterisk to SPA941 has some codec bits to work out, I
think its to do with asterisk PCMU being different ? I know I had something
similar when trying to use the same setup with gizmo (which I fixed by
forcing G729 from SPA941 -> asterisk -> freeswitch.

Cheers

Kirk


> Date: Wed, 17 Dec 2008 11:21:32 -0600
> From: Brian West <brian at freeswitch.org>
> Subject: Re: [Freeswitch-users] Call sip phones from gtalk / jabber
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <876638A9-FBB0-4079-A76F-814E6953D395 at freeswitch.org>
> Content-Type: text/plain; charset="us-ascii"
>
> FreeSWITCH already logs into your jabber server as a component if you
> cant communicate with other domains then your jabber server is not
> configured correctly.
>
> /b
>
> On Dec 17, 2008, at 11:15 AM, Kirk Bateman wrote:
>
> > Kriko,
> >
> > I have been looking at the same sort of thing, but I'm planning to
> > implement an ejabberd bot component (so I can hopefully use the new
> > mod_erlang_event freeswitch interface).
> >
> > It seems to me that bits of the current dingaling / jingle interface
> > are having problems, like not liking sending messages to other
> > domains, its generally working if they are all .gmail.com users but
> > when you have some ongooglemail.com etc it starts breaking, and
> > doesn't use the whole JID in the from attribute for sending messages.
> >
> > When I get a chance I'll try and narrow down the problem.
> >
> > Cheers
> >
> > Kirk
>
>
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From fidibus83 at aol.com  Thu Dec 18 01:53:42 2008
From: fidibus83 at aol.com (fidibus83)
Date: Thu, 18 Dec 2008 10:53:42 +0100
Subject: [Freeswitch-users] SQL Error
In-Reply-To: <200812182237.33739.hads@nice.net.nz>
References: <004801c960f1$dc680840$6445310a@Franzi>
	<200812182237.33739.hads@nice.net.nz>
Message-ID: <006b01c960f6$83584030$6445310a@Franzi>

Thanks. It's ok again!

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Hadley
Rich
Gesendet: Donnerstag, 18. Dezember 2008 10:38
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] SQL Error

On Thursday 18 December 2008 22:20:24 fidibus83 wrote:
> I?m a newbie in FS. I get an error from the freeswitch cli:
>
> 2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL
> ERR [database disk image is malformed]
>
> I don?t know what to do to remove this error! Can you help me?

If you remove the database files (which are in $PREFIX/db) then FreeSWITCH
will 
recreate then at startup.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From kirk.bateman at gmail.com  Thu Dec 18 02:03:00 2008
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Thu, 18 Dec 2008 10:03:00 +0000
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
Message-ID: <2bee4fc40812180203u390f189ev970a7c714662185a@mail.gmail.com>

Kriko,

Have a look at this, I used it to get my gtalk to fs working.

http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/

Cheers

Kirk

Date: Thu, 18 Dec 2008 09:37:38 +0100
> From: kriko <kristjan.ugrin at gmail.com>
> Subject: Re: [Freeswitch-users] Call sip phones from gtalk / jabber
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <op.umcnkzr2atcbto at linux.site>
> Content-Type: text/plain; format=flowed; delsp=yes; charset=utf-8
>
> I'm not sure if we understood each other correctly.
> I meant calling from jabber to other sip phones. I'm not sure you can just
> add a sip phone as
> a buddy into e.g. gtalk or any other jabber service and call it.
> So that why a bot (and only one but), which you would have as a buddy and
> you would feed him with numbers.
> And he would estabilish a call between jabber user and typed in number.
> Was I clearer?
>
> I'll take a look how the component mode works, hopefully there is more
> documentation than only this:
> http://wiki.freeswitch.org/wiki/Dingaling
>
> Also if this functionality is not possible with fs as it is, I could maybe
> write a java program which interact with fs
> interface and do that as intended?
>
> Cheers
>
>
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From kristjan.ugrin at gmail.com  Thu Dec 18 02:39:13 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 18 Dec 2008 11:39:13 +0100
Subject: [Freeswitch-users] Java example
Message-ID: <op.umcs7ncwatcbto@linux.site>

I made a simple java example, following this guide
http://wiki.freeswitch.org/wiki/Java

so when someone calls it should print something in console.
I've also modified dialplan/public.xml, what I want is to intercept calls  
 from jabber and process them:
http://pastebin.com/m35de11d9

But there is nothing printed into console, path to jar is ok, also class  
name is ok.
Java module is being loaded on startup.

What could be wrong?

-- 
kriko



From damjan at ecntelecoms.com  Thu Dec 18 02:53:22 2008
From: damjan at ecntelecoms.com (damjan at ecntelecoms.com)
Date: Thu, 18 Dec 2008 12:53:22 +0200 (SAST)
Subject: [Freeswitch-users] Java example
In-Reply-To: <op.umcs7ncwatcbto@linux.site>
References: <op.umcs7ncwatcbto@linux.site>
Message-ID: <6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>

> I made a simple java example, following this guide
> http://wiki.freeswitch.org/wiki/Java
>
> so when someone calls it should print something in console.
> I've also modified dialplan/public.xml, what I want is to intercept calls
>  from jabber and process them:
> http://pastebin.com/m35de11d9
>
> But there is nothing printed into console, path to jar is ok, also class
> name is ok.
> Java module is being loaded on startup.
>
> What could be wrong?
>
> --
> kriko

You need to specify the path to the JAR file containing
si.marand.freeswitch.PhoneTest, which is definitely not freeswitch.jar.

Otherwise try attaching a remote debugger to the Java module and trace
through it as described on that wiki.

Damjan





From kristjan.ugrin at gmail.com  Thu Dec 18 04:09:48 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 18 Dec 2008 13:09:48 +0100
Subject: [Freeswitch-users] Java example
In-Reply-To: <6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>
References: <op.umcs7ncwatcbto@linux.site>
	<6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>
Message-ID: <op.umcxenzsatcbto@linux.site>

It is the right jar, I renamed it now to phoneTest.jar but still not  
working.
Do I have to specify whole path to the program inside class or just the  
class?
Remote debugging is working, but breakpoints never got triggered, so it is  
not being
executed at all.

I'm trying to process a call from gtalk to sip, maybe my public.xml is  
misconfigured (now I changed jar to phoneTest.jar):
http://pastebin.com/m35de11d9
(or public.xml it is not the right place at all?)


On Thu, 18 Dec 2008 11:53:22 +0100, <damjan at ecntelecoms.com> wrote:

> You need to specify the path to the JAR file containing
> si.marand.freeswitch.PhoneTest, which is definitely not freeswitch.jar.
>
> Otherwise try attaching a remote debugger to the Java module and trace
> through it as described on that wiki.
>
> Damjan
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
kriko



From mike at jerris.com  Thu Dec 18 05:10:19 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Dec 2008 08:10:19 -0500
Subject: [Freeswitch-users] Question about running Freeswitch in the
	background
In-Reply-To: <20081218082907.GA11459@jdc.jasonjgw.net>
References: <20ad6b920812180001t2ef2a2c7occ8130c57c69f3f5@mail.gmail.com>
	<20081218082907.GA11459@jdc.jasonjgw.net>
Message-ID: <74FAFF44-C6F1-4AF8-A0C9-12DCDD16992C@jerris.com>

A note, logging is handled by mod_logfile, it has nothing to do if you  
run in the background or not.

Mike

On Dec 18, 2008, at 3:29 AM, Jason White wrote:

> On Thu, Dec 18, 2008 at 04:01:07PM +0800, mark morreny wrote:
>> I have a small question about running Freeswich in the background  
>> using -nc
>> option.  Does freeswitch writes the log in any log file when  
>> running in
>> background mode?
>
> Yes. Have a look at ~freeswitch/log/freeswitch.log
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Thu Dec 18 05:12:03 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Dec 2008 08:12:03 -0500
Subject: [Freeswitch-users] Call sip phones from gtalk / jabber
In-Reply-To: <2bee4fc40812180145mda36e7jf1b2905b9a5c63af@mail.gmail.com>
References: <2bee4fc40812180145mda36e7jf1b2905b9a5c63af@mail.gmail.com>
Message-ID: <F3079C63-BB3B-4487-B458-2B941B49E91C@jerris.com>


On Dec 18, 2008, at 4:45 AM, Kirk Bateman wrote:

>  have had Freeswitch connecting to GTalk directly as a client and  
> that was where I was getting the issues with sending anything to  
> other domains.

I have seen this before specifically with gmail being unable to  
federate presense... their stuff can be really flakey, do you see the  
same thing if you client login to a jabber.org address?




From yudha2008 at gmail.com  Thu Dec 18 05:13:08 2008
From: yudha2008 at gmail.com (Baskar)
Date: Thu, 18 Dec 2008 18:43:08 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
Message-ID: <d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>

*Hi,

I am using JavaScript file to detect busy tone signals but I cant able to
detect the busy tone signals
*
*My JavaScript*
*
session1 = new Session();
session1.originate(session1, "{ignore_early_media=true}sofia/default/
39841799874 at 172.20.191.228");
session1.execute("tone_detect", "busy 400 r");
session1.execute("bridge", "sofia/default/39841799874 at 172.20.191.228");
session1.execute("transfer", "39841799874");*


*I get output:*

*freeswitch at localhost.localdomain> jsrun tone.js*
API CALL [jsrun(tone.js)] output:
OK

freeswitch at localhost.localdomain> 2008-12-18 18:42:30 [NOTICE]
switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/
39841799874 at 172.20.191.228 [0a9723ca-d170-4cee-a8bf-58a8ad018a44]
2008-12-18 18:42:35 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Ring-Ready sofia/internal/39841799874 at 172.20.191.228!
2008-12-18 18:42:35 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Pre-Answer sofia/internal/39841799874 at 172.20.191.228!
2008-12-18 18:42:39 [NOTICE] sofia.c:2963 sofia_handle_sip_i_state() Channel
[sofia/internal/39841799874 at 172.20.191.228] has been answered
2008-12-18 18:42:39 [NOTICE] mod_dptools.c:1217
tone_detect_session_function()* Enabling tone detection 'busy' '400'*
2008-12-18 18:42:39 [NOTICE] switch_channel.c:565 switch_channel_set_name()
New Channel sofia/internal/39841799874 at 172.20.191.228[5152416f-7e5c-4a60-9601-6a4af625d8aa]
2008-12-18 18:42:39 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Ring-Ready sofia/internal/39841799874 at 172.20.191.228!
2008-12-18 18:42:39 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media()
Pre-Answer sofia/internal/39841799874 at 172.20.191.228!
*
when i run my js call is connected and after the caller answer only enabling
the tone detection.

I am not sure i am correct. correct me how to detect the busy signal. I have
written a small JavaScript. Correct me where i am wrong (In the program or
in the way it detect the call).

Thanks in advance.
*
*--
Warm Regards,
N.Baskar
*
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From anthony.minessale at gmail.com  Thu Dec 18 06:12:09 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Dec 2008 08:12:09 -0600
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
In-Reply-To: <4949B946.5050502@novatex.com.au>
References: <49478B0F.3000802@novatex.com.au>
	<153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>
	<4949B946.5050502@novatex.com.au>
Message-ID: <191c3a030812180612n94f9b72vc9ad0ba2d90d6a9e@mail.gmail.com>

can you press f8 to set the FS console to DEBUG and take the same capture.


On Wed, Dec 17, 2008 at 8:45 PM, Scott Ellis <scott.ellis at novatex.com.au>wrote:

>  After further checking, it does not seem like the authentication after the
> challenge is being sent...
>
> Are there any other settings I should be aware of other than placing the
> file in external and setting register to true?
>
> Scott
>
> 2008-12-18 13:32:28 [NOTICE] sofia_reg.c:265 sofia_reg_check_gateway()
> Registering sip.pennytel.com
> nua: nua_handle_bind: entering
> nua: nua_register: entering
> nua(0x89b08e0): sent signal r_register
> nua(0x89b08e0): recv signal r_register
> nua: nua_stack_set_params: entering
> soa_clone(static::0x8977798, 0x89792f8, 0x89b08e0) called
> soa_set_params(static::0x89c52c0, ...) called
> soa_set_params(static::0x89c52c0, ...) called
> nta_leg_tcreate(0x89c4948)
> nua(0x89b08e0): adding register usage
> nta: selecting scheme sip
> nta: for "sip.pennytel.com" query "_sip._udp.sip.pennytel.com" SRV
> nta: for "sip.pennytel.com" query "sip.pennytel.com" A (cached)
> nta: sip.pennytel.com. IN A 202.85.243.87
> tport_tsend(0x8976740) tpn = udp/202.85.243.87:5060
> tport_resolve addrinfo = 202.85.243.87:5060
> tport_by_addrinfo(0x8976740): not found by name udp/202.85.243.87:5060
> tport_vsend(0x8976740): 646 bytes of 646 to udp/202.85.243.87:5060
> tport_vsend returned 646
> send 646 bytes to udp/[202.85.243.87]:5060 at 02:32:30.322198:
>    ------------------------------------------------------------------------
>    REGISTER sip:sip.pennytel.com;transport=udp SIP/2.0
>    Via: SIP/2.0/UDP 203.113.255.140:5080;rport;branch=z9hG4bKt232eUFUNXr2e
>    Max-Forwards: 70
>    From: <sip:8xxxxx75 at sip.pennytel.com;transport=udp>;tag=t0Umc83St29ND
>    To: <sip:8xxxxxx75 at sip.pennytel.com;transport=udp>
>    Call-ID: d25d6f36-ccab-11dd-900f-67e92a02be7d
>    CSeq: 108665407 REGISTER
>    Contact: <sip:8xxxxxxx75 at 203.113.255.140:5080;transport=udp>
>    Expires: 600
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10760
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO
>    Supported: timer, precondition, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> nta: sent REGISTER (108665407) to udp/202.85.243.87:5060
> tport_pend(0x8976740): pending 0x89f2d50 for udp/192.168.0.5:5080 (already
> 0)
> nta: timer set to 32000 ms
> nta: timer shortened to 500 ms
> tport_wakeup_pri(0x8976740): events IN
> tport_recv_event(0x8976740)
> tport_recv_iovec(0x8976740) msg 0x89eeeb8 from (udp/192.168.0.5:5080) has
> 518 bytes, veclen = 1
> recv 518 bytes from udp/[202.85.243.87]:5060 at 02:32:30.370072:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP 203.113.255.140:5080
> ;rport=5080;branch=z9hG4bKt232eUFUNXr2e
>    From: <sip:8xxxxx75 at sip.pennytel.com;transport=udp>;tag=t0Umc83St29ND
>    To: <sip:8xxxxx75 at sip.pennytel.com;transport=udp>
> ;tag=abda4710fbd488d9ce6d01bba5c3e23b-cec7
>    Call-ID: d25d6f36-ccab-11dd-900f-67e92a02be7d
>    CSeq: 108665407 REGISTER
>    WWW-Authenticate: Digest realm="sip.pennytel.com",
> nonce="4949b76bf622961d78acb213b5556104938ecd6e"
>    Server: Sip EXpress router (0.9.6 (i386/freebsd))
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> tport_deliver(0x8976740): msg 0x89eeeb8 (518 bytes) from udp/
> 202.85.243.87:5080/sip next=(nil)
> nta: received 401 Unauthorized for REGISTER (108665407)
> nta: 401 Unauthorized is going to a transaction
> nta_outgoing: RTT is 49.89 ms
> tport_release(0x8976740): 0x89f2d50 by 0x89a4640 with 0x89eeeb8
> nta: timer set next to 4531 ms
> nta: timer K fired, terminate REGISTER (108665407)
> outgoing_reclaim_all((nil), (nil), 0xb2c6d1e8)
> nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/2 free
> nta: timer not set
> 2008-12-18 13:32:59 [WARNING] sofia_reg.c:307 sofia_reg_check_gateway()
> sip.pennytel.com Failed Registration, setting retry to 60 seconds.
>
>
> Michael Jerris wrote:
>
> We send authentication after we get a challenge because on startup we
> need the nonce from them to build the hash in the Auth header properly.
>
> Mike
>
> On Dec 16, 2008, at 6:03 AM, Scott Ellis wrote:
>
>
>
>  I have a standard install, and I am trying to get a Pennytel gateway
> to
> register.
>
> After looking at Wireshark traces of x-lite registering and FreeSwitch
> registering, FreeSwitch is not sending any authentication information
> with the registration request. I am obviously missing something here!
>
> I understand for incoming calls you don't want authentication, but for
> outgoing it is obviously required.
>
> Is there a flag somewhere that I am supposed to set? The file was
> taken
> from the wiki page, and looks like it was previously tested when using
> the obsolete outbound directory structure.
>
> The following file is in the conf/sip_profiles/external directory.
>
> <include>
>  <gateway name="PennyTel">
>    <param name="username" value="8xxxxxxx"/>
>    <param name="password" value="xxxxxxx"/>
>    <param name="realm" value="sip.pennytel.com"/>
>    <param name="proxy" value="sip.pennytel.com"/>
>    <param name="register" value="true"/>
>    <param name="expire-seconds" value="60"/>
>  </gateway>
> </include>
>
> Thanks.
>
> Scott
>
>
> _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>  _______________________________________________
> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Dec 18 06:19:22 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Dec 2008 08:19:22 -0600
Subject: [Freeswitch-users] Java example
In-Reply-To: <op.umcxenzsatcbto@linux.site>
References: <op.umcs7ncwatcbto@linux.site>
	<6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>
	<op.umcxenzsatcbto@linux.site>
Message-ID: <191c3a030812180619k394a9e33g7cba5808e7d95558@mail.gmail.com>

did you turn up your console log level high enough to see it? The default
level is "INFO"


On Thu, Dec 18, 2008 at 6:09 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> It is the right jar, I renamed it now to phoneTest.jar but still not
> working.
> Do I have to specify whole path to the program inside class or just the
> class?
> Remote debugging is working, but breakpoints never got triggered, so it is
> not being
> executed at all.
>
> I'm trying to process a call from gtalk to sip, maybe my public.xml is
> misconfigured (now I changed jar to phoneTest.jar):
> http://pastebin.com/m35de11d9
> (or public.xml it is not the right place at all?)
>
>
> On Thu, 18 Dec 2008 11:53:22 +0100, <damjan at ecntelecoms.com> wrote:
>
> > You need to specify the path to the JAR file containing
> > si.marand.freeswitch.PhoneTest, which is definitely not freeswitch.jar.
> >
> > Otherwise try attaching a remote debugger to the Java module and trace
> > through it as described on that wiki.
> >
> > Damjan
> >
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> kriko
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From peder at networkoblivion.com  Thu Dec 18 06:38:34 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Thu, 18 Dec 2008 08:38:34 -0600
Subject: [Freeswitch-users] Core Dump
Message-ID: <494A606A.2000601@networkoblivion.com>

What is the process for capturing and submitting a core dump?

I am messing around with the Cisco 79x1 phones and tcp and multiple reg. 
  I have a 7961 using tcp and a 7960 using udp both reg'd with the same 
number and both showing up as registered.  If I call out from the phone 
using tcp, it works.  If I call out from the phone using udp, I get a 
core dump.  If I call in, it calls both phones and I am assuming the 
call to the phone using udp causes a core dump as well.  These are the 
only two phones on the system and I am running version 10851 from 
yesterday.  If I only have the udp phone registered and the tcp phone is 
off, it works fine.  It is only when I have a mix of a udp reg and a tcp 
reg on the same number that I appear to get a core dump.


Peder



From msc at freeswitch.org  Thu Dec 18 06:43:06 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 18 Dec 2008 06:43:06 -0800
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
Message-ID: <980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>

You've got ignore_early_media set to true but busy signals might be  
sent during early media. Why are you ignoring early media?

Also, you might need to check your tone_detect syntax. You're set to  
detect 400Hz but you haven't told the system what to do if it does  
detect that tone. Please look at the wiki examples for tone_detect.  
You will see what kinds of things you can do with it, but usually you  
just set a channel variable.

-MC

Sent from my iPhone

On Dec 18, 2008, at 5:13 AM, Baskar <yudha2008 at gmail.com> wrote:

> Hi,
>
> I am using JavaScript file to detect busy tone signals but I cant  
> able to detect the busy tone signals
>
> My JavaScript
>
> session1 = new Session();
> session1.originate(session1, "{ignore_early_media=true}sofia/default/39841799874 at 172.20.191.228 
> ");
> session1.execute("tone_detect", "busy 400 r");
> session1.execute("bridge", "sofia/default/ 
> 39841799874 at 172.20.191.228");
> session1.execute("transfer", "39841799874");
>
>
> I get output:
>
> freeswitch at localhost.localdomain> jsrun tone.js
> API CALL [jsrun(tone.js)] output:
> OK
>
> freeswitch at localhost.localdomain> 2008-12-18 18:42:30 [NOTICE]  
> switch_channel.c:565 switch_channel_set_name() New Channel sofia/ 
> internal/39841799874 at 172.20.191.228 [0a9723ca-d170-4cee- 
> a8bf-58a8ad018a44]
> 2008-12-18 18:42:35 [NOTICE] sofia_glue.c:2097  
> sofia_glue_tech_media() Ring-Ready sofia/internal/39841799874 at 172.20.191.228 
> !
> 2008-12-18 18:42:35 [NOTICE] sofia_glue.c:2097  
> sofia_glue_tech_media() Pre-Answer sofia/internal/39841799874 at 172.20.191.228 
> !
> 2008-12-18 18:42:39 [NOTICE] sofia.c:2963 sofia_handle_sip_i_state()  
> Channel [sofia/internal/39841799874 at 172.20.191.228] has been answered
> 2008-12-18 18:42:39 [NOTICE] mod_dptools.c:1217  
> tone_detect_session_function() Enabling tone detection 'busy' '400'
> 2008-12-18 18:42:39 [NOTICE] switch_channel.c:565  
> switch_channel_set_name() New Channel sofia/internal/39841799874 at 172.20.191.228 
>  [5152416f-7e5c-4a60-9601-6a4af625d8aa]
> 2008-12-18 18:42:39 [NOTICE] sofia_glue.c:2097  
> sofia_glue_tech_media() Ring-Ready sofia/internal/39841799874 at 172.20.191.228 
> !
> 2008-12-18 18:42:39 [NOTICE] sofia_glue.c:2097  
> sofia_glue_tech_media() Pre-Answer sofia/internal/39841799874 at 172.20.191.228 
> !
>
> when i run my js call is connected and after the caller answer only  
> enabling the tone detection.
>
> I am not sure i am correct. correct me how to detect the busy  
> signal. I have written a small JavaScript. Correct me where i am  
> wrong (In the program or in the way it detect the call).
>
> Thanks in advance.
>
> -- 
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From fidibus83 at aol.com  Thu Dec 18 06:43:19 2008
From: fidibus83 at aol.com (fidibus83)
Date: Thu, 18 Dec 2008 15:43:19 +0100
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21069519.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com>
	<49490301.8040701@lists.rupa.com><21054080.post@talk.nabble.com>
	<49491847.3030403@lists.rupa.com> <21069519.post@talk.nabble.com>
Message-ID: <012501c9611e$f8e43bb0$6445310a@Franzi>

Hello,

Carole, your conference-programm is what I looked for. It's great. I try it.
But I get an Error, when I press *1 and I don't know what it mean... Have
you an idea?

[ERR] mod_conference.c:4849 conference_new() invalid Record! No name.
[CRIT] mod_conference.c:4314 conference_function() Memory Error!

Thanks! 

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Carole
O.
Gesendet: Donnerstag, 18. Dezember 2008 10:38
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] dynamic conference


Hello,

Thanks for your answers!
Concerning the creation of a new variable for the conference the problem is
that I do not create channels from the conference. I call separately a new
member on a new channel and add it on the conference only if he agrees to
enter it. So it was the same problem as for the uuid, I am not sure I can
access the good variable from anywhere in case many conferences are running.

So, I do the following if somebody is interested in:

<extension name="Local_Extension">
     .....................
      <condition field="destination_number"
expression="^${caller_id_number}$">
        <anti-action application="bind_meta_app" data="1 a o transfer::
transf_opp_leg_${db(select/conference/${caller_id_number})}_${destination_nu
mber}"/>
        <anti-action application="bind_meta_app" data="2 ab s
transfer::-both
transf_both_${uuid}_${caller_id_number}_${destination_number}"/>
       ..........
</extension>

<extension name="transf_both_legs">
     <condition field="destination_number"
expression="^transf_both_(.*)_(.*)_(.*)$">
         <action application="db" data="insert/conference/$2/conf_$1"/>
         <action application="db" data="insert/conference/$3/conf_$1"/>
         <action application="answer"/>
         <action application="conference" data="conf_$1 at conf_dynamic"/>
     </condition>
</extension>

<extension name="transf_opposite_leg">
     <condition field="destination_number"
expression="^transf_opp_leg_(.*)_(.*)$">
         <action application="db" data="insert/conference/$2/$1"/>
         <action application="answer"/>
         <action application="conference" data="$1 at conf_dynamic"/>
     </condition>
</extension>

This is not perfect and I am not sure it is "cleanly" programmed but it
works and it is flexible. All the members in the conference can invite new
members thanks to the conference name they all have in the database.

If you still have critics they are all welcome!
Thanks for your help,
Best regards,
Carole


Rupa Schomaker (lists)-2 wrote:
> 
> On 12/17/2008 8:24 AM, Carole O. wrote:
>> It would be unique you are right but I am not sure I can get its value if
>> A
>> puts the call on hold, calls C and wants to add it to the conference
>> whose
>> name dependent of the uuid of another session. 
>> I think if I use ${uuid} to add C I will have the uuid of the session
>> between A and C and not A and B no?
>> And I really have to configure this from the dialplan so statically.
>> 
>> Am I wrong somewhere??
>> 
>> Carole
> 
> Ah, yeah.  uuid would not be the same when initiating a new call that
> you then transfer to the conference call.  You need something that is
> intrinsic to the endpoint.
> 
> I did a quick info dump to an originated call.  Depending on your
> use-case (are these calls originating from registered handsets, trunked
> from a sip provider, etc) you might want to rely on the variable
> "sip_contact_uri" which is a combination of registered user name and ip
> (and port if port isn't 5060).  This should be unique per endpoint.
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context:
http://www.nabble.com/dynamic-conference-tp21053181p21069519.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From kristjan.ugrin at gmail.com  Thu Dec 18 06:56:11 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 18 Dec 2008 15:56:11 +0100
Subject: [Freeswitch-users] Gtalk to sip problems when reconfiguring from
	scratch
Message-ID: <op.umc43xs3atcbto@linux.site>

I recently purged all freeswitch config and restarted configuring from  
scratch. Using defaults, I modified public.xml dialplan config (added line  
16 - 28):
http://pastebin.com/m5ece6e6f

and added a new config under jingle_profiles:
http://pastebin.com/d6e983b99

I register with phonelite or twinkle as user 1000 (it says successfull  
reg.), but when I make a call from gtalk I hear that user is not  
available, none of
client rings.

When registering sofia prints:
2008-12-18 15:53:32 [INFO] sofia_presence.c:475  
actual_sofia_presence_event_handler() internal START_PRESENCE_PROBE_SQL
2008-12-18 15:53:32 [NOTICE] sofia_presence.c:793  
sofia_presence_resub_callback() internal PRESENCE_PROBE 1000 at 10.99.8.221
2008-12-18 15:53:32 [INFO] sofia_presence.c:484  
actual_sofia_presence_event_handler() internal END_PRESENCE_PROBE_SQL

2008-12-18 15:53:32 [INFO] sofia_presence.c:547  
actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal)
2008-12-18 15:53:32 [INFO] sofia_presence.c:563  
actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal)
2008-12-18 15:53:32 [WARNING] sofia_presence.c:517  
actual_sofia_presence_event_handler() external is passive, skipping
2008-12-18 15:53:32 [WARNING] sofia_presence.c:510  
actual_sofia_presence_event_handler() 10.99.8.221 is an alias, skipping
2008-12-18 15:53:32 [WARNING] sofia_presence.c:510  
actual_sofia_presence_event_handler() default is an alias, skipping
2008-12-18 15:53:32 [WARNING] sofia_presence.c:510  
actual_sofia_presence_event_handler() nat is an alias, skipping
2008-12-18 15:53:32 [WARNING] sofia_presence.c:510  
actual_sofia_presence_event_handler() outbound is an alias, skipping

and when fetching registrations:
2008-12-18 15:53:50 [ERR] sofia_reg.c:1120  
sofia_reg_handle_sip_i_register() NO CONTACT!

Does this means that registration failed? Why it doesn't call anymore the  
targeted user (1000)?

Here is also an extract from console log while calling, it caught my  
attention:
2008-12-18 15:54:41 [WARNING] mod_dptools.c:2047 user_outgoing_channel()  
Can't find user [1000@]
2008-12-18 15:54:41 [ERR] switch_ivr_originate.c:1110  
switch_ivr_originate() Cannot create outgoing channel of type [user]  
cause: [SUBSCRIBER_ABSENT]
2008-12-18 15:54:41 [DEBUG] switch_ivr_originate.c:1689  
switch_ivr_originate() Originate Resulted in Error Cause: 20  
[SUBSCRIBER_ABSENT]

-- 
kriko



From msc at freeswitch.org  Thu Dec 18 07:02:18 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 18 Dec 2008 07:02:18 -0800
Subject: [Freeswitch-users] Core Dump
Message-ID: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>

Check out this page:
wiki.freeswitch.org/wiki/Debugging_Freeswitch

-MC

Sent from my iPhone

On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com 
 > wrote:

> What is the process for capturing and submitting a core dump?
>
> I am messing around with the Cisco 79x1 phones and tcp and multiple  
> reg.
> I have a 7961 using tcp and a 7960 using udp both reg'd with the same
> number and both showing up as registered.  If I call out from the  
> phone
> using tcp, it works.  If I call out from the phone using udp, I get a
> core dump.  If I call in, it calls both phones and I am assuming the
> call to the phone using udp causes a core dump as well.  These are the
> only two phones on the system and I am running version 10851 from
> yesterday.  If I only have the udp phone registered and the tcp  
> phone is
> off, it works fine.  It is only when I have a mix of a udp reg and a  
> tcp
> reg on the same number that I appear to get a core dump.
>
>
> Peder
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From peder at networkoblivion.com  Thu Dec 18 07:36:00 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Thu, 18 Dec 2008 09:36:00 -0600
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
Message-ID: <494A6DE0.4030701@networkoblivion.com>

If anybody wants to look at the core dump in gdb, here it is (the actual 
core is 256Meg):

http://pastebin.freeswitch.org/6476

I know zip about debugging and gdb, but from looking through it, I see a 
segmentation fault and it appears to be thread 15094.  The last three 
items in the bt full for that thread are:

destroy_status = <value optimized out>
fd = (switch_file_t *) 0x80529b0
pool = (switch_memory_pool_t *) 0x80528f0

So I would guess it is trying to access an invalid memory location, but 
why, I have no idea....

Any ideas?


Peder


Michael S Collins wrote:
> Check out this page:
> wiki.freeswitch.org/wiki/Debugging_Freeswitch
> 
> -MC
> 
> Sent from my iPhone
> 
> On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com 
>  > wrote:
> 
>> What is the process for capturing and submitting a core dump?
>>
>> I am messing around with the Cisco 79x1 phones and tcp and multiple  
>> reg.
>> I have a 7961 using tcp and a 7960 using udp both reg'd with the same
>> number and both showing up as registered.  If I call out from the  
>> phone
>> using tcp, it works.  If I call out from the phone using udp, I get a
>> core dump.  If I call in, it calls both phones and I am assuming the
>> call to the phone using udp causes a core dump as well.  These are the
>> only two phones on the system and I am running version 10851 from
>> yesterday.  If I only have the udp phone registered and the tcp  
>> phone is
>> off, it works fine.  It is only when I have a mix of a udp reg and a  
>> tcp
>> reg on the same number that I appear to get a core dump.
>>
>>
>> Peder
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From msc at freeswitch.org  Thu Dec 18 08:09:50 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 18 Dec 2008 08:09:50 -0800
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <494A6DE0.4030701@networkoblivion.com>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<494A6DE0.4030701@networkoblivion.com>
Message-ID: <CAEDC248-D670-4603-A4A8-5F06074304E9@freeswitch.org>

Is this a single occurrence or can you make it happen consistently?
-MC

Sent from my iPhone

On Dec 18, 2008, at 7:36 AM, "peder at networkoblivion.com" <peder at networkoblivion.com 
 > wrote:

> If anybody wants to look at the core dump in gdb, here it is (the  
> actual
> core is 256Meg):
>
> http://pastebin.freeswitch.org/6476
>
> I know zip about debugging and gdb, but from looking through it, I  
> see a
> segmentation fault and it appears to be thread 15094.  The last three
> items in the bt full for that thread are:
>
> destroy_status = <value optimized out>
> fd = (switch_file_t *) 0x80529b0
> pool = (switch_memory_pool_t *) 0x80528f0
>
> So I would guess it is trying to access an invalid memory location,  
> but
> why, I have no idea....
>
> Any ideas?
>
>
> Peder
>
>
> Michael S Collins wrote:
>> Check out this page:
>> wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com
>>> wrote:
>>
>>> What is the process for capturing and submitting a core dump?
>>>
>>> I am messing around with the Cisco 79x1 phones and tcp and multiple
>>> reg.
>>> I have a 7961 using tcp and a 7960 using udp both reg'd with the  
>>> same
>>> number and both showing up as registered.  If I call out from the
>>> phone
>>> using tcp, it works.  If I call out from the phone using udp, I  
>>> get a
>>> core dump.  If I call in, it calls both phones and I am assuming the
>>> call to the phone using udp causes a core dump as well.  These are  
>>> the
>>> only two phones on the system and I am running version 10851 from
>>> yesterday.  If I only have the udp phone registered and the tcp
>>> phone is
>>> off, it works fine.  It is only when I have a mix of a udp reg and a
>>> tcp
>>> reg on the same number that I appear to get a core dump.
>>>
>>>
>>> Peder
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Dec 18 08:21:11 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 10:21:11 -0600
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <494A6DE0.4030701@networkoblivion.com>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<494A6DE0.4030701@networkoblivion.com>
Message-ID: <C9C3FF52-4FA8-4241-8DDB-7672F194162D@freeswitch.org>

Peder,
	Can you join us on IRC.
/b

On Dec 18, 2008, at 9:36 AM, peder at networkoblivion.com wrote:

> If anybody wants to look at the core dump in gdb, here it is (the  
> actual
> core is 256Meg):
>
> http://pastebin.freeswitch.org/6476
>
> I know zip about debugging and gdb, but from looking through it, I  
> see a
> segmentation fault and it appears to be thread 15094.  The last three
> items in the bt full for that thread are:
>
> destroy_status = <value optimized out>
> fd = (switch_file_t *) 0x80529b0
> pool = (switch_memory_pool_t *) 0x80528f0
>
> So I would guess it is trying to access an invalid memory location,  
> but
> why, I have no idea....
>
> Any ideas?
>
>
> Peder
>
>
> Michael S Collins wrote:
>> Check out this page:
>> wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com
>>> wrote:
>>
>>> What is the process for capturing and submitting a core dump?
>>>
>>> I am messing around with the Cisco 79x1 phones and tcp and multiple
>>> reg.
>>> I have a 7961 using tcp and a 7960 using udp both reg'd with the  
>>> same
>>> number and both showing up as registered.  If I call out from the
>>> phone
>>> using tcp, it works.  If I call out from the phone using udp, I  
>>> get a
>>> core dump.  If I call in, it calls both phones and I am assuming the
>>> call to the phone using udp causes a core dump as well.  These are  
>>> the
>>> only two phones on the system and I am running version 10851 from
>>> yesterday.  If I only have the udp phone registered and the tcp
>>> phone is
>>> off, it works fine.  It is only when I have a mix of a udp reg and a
>>> tcp
>>> reg on the same number that I appear to get a core dump.
>>>
>>>
>>> Peder
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From odermann at googlemail.com  Thu Dec 18 08:36:32 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 17:36:32 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
Message-ID: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>

i would like to know, what the best way is, to redirect an incoming
call from one fs (fs1) to another fs (fs2).

we use 3 freeswitch servers and the carrier passes calls to the three
fs servers randomly. if on fs server is not offline, the carrier sends
the call to the next fs.
this is generally good, but for conferencing it not so good.

i am using socket outbound and need to do this for conferencing. let's
say, we have a conference going on on fs1. another person wants to
enter this conference, but the call is passed to fs2. on fs2 we see,
that the caller wants to enter the conference going on on fs1.

now we have to redirect the call from fs2 to fs1. is this done with
"redirect" and some according settings/params or are there other ways
to do this? we would like to do this without our carrier doing
something, to be a little more independant.

thanks
dennis



From brian at freeswitch.org  Thu Dec 18 08:43:25 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 10:43:25 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
Message-ID: <283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>

the deflect app.

/b

On Dec 18, 2008, at 10:36 AM, Dennis wrote:

> i would like to know, what the best way is, to redirect an incoming
> call from one fs (fs1) to another fs (fs2).




From odermann at googlemail.com  Thu Dec 18 08:52:23 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 17:52:23 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
Message-ID: <5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>

i had a look at the deflect app, but as far as i understand it, the
carrier has to support/understand it ans react on the signals.

is that right or does this have nothing to do with our carrier? or
does this work between fs servers in the same local network?


another similar question is: how to reject calls, so that the carrier
tries to route the call to another fs? if we want to make changes to
fs and test them, we would like to block new incoming calls, till
there are no running calls, to shut down the fs.




2008/12/18 Brian West <brian at freeswitch.org>:
> the deflect app.
>
> /b
>
> On Dec 18, 2008, at 10:36 AM, Dennis wrote:
>
>> i would like to know, what the best way is, to redirect an incoming
>> call from one fs (fs1) to another fs (fs2).
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Thu Dec 18 08:58:24 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 10:58:24 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
Message-ID: <704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>

What switch is your provider using?

/b

On Dec 18, 2008, at 10:52 AM, Dennis wrote:

> i had a look at the deflect app, but as far as i understand it, the
> carrier has to support/understand it ans react on the signals.
>
> is that right or does this have nothing to do with our carrier? or
> does this work between fs servers in the same local network?
>
>
> another similar question is: how to reject calls, so that the carrier
> tries to route the call to another fs? if we want to make changes to
> fs and test them, we would like to block new incoming calls, till
> there are no running calls, to shut down the fs.




From jonas.gauffin at gmail.com  Thu Dec 18 09:05:06 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 18 Dec 2008 18:05:06 +0100
Subject: [Freeswitch-users] "choppy" voice
Message-ID: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>

Hello

I got problems with choppy voice. I just happens1 time of 10 or something
like that.
Incorrect call: http://pastebin.freeswitch.org/6479
working call: http://pastebin.freeswitch.org/6481

Any idea what is wrong?

Thanks,
  Jonas
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From odermann at googlemail.com  Thu Dec 18 09:07:20 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 18:07:20 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
Message-ID: <5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>

sorry, i do not know that. i could ask tomorrow.

is deflect, what i understand? the provider has to support it? if yes,
what could i tell and ask the provider, to find a solution to this
problem? the provider is quite open for new ideas, although we do not
want to be to dependant on the provider and his possibilities.


2008/12/18 Brian West <brian at freeswitch.org>:
> What switch is your provider using?
>
> /b
>
> On Dec 18, 2008, at 10:52 AM, Dennis wrote:
>
>> i had a look at the deflect app, but as far as i understand it, the
>> carrier has to support/understand it ans react on the signals.
>>
>> is that right or does this have nothing to do with our carrier? or
>> does this work between fs servers in the same local network?
>>
>>
>> another similar question is: how to reject calls, so that the carrier
>> tries to route the call to another fs? if we want to make changes to
>> fs and test them, we would like to block new incoming calls, till
>> there are no running calls, to shut down the fs.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Thu Dec 18 09:19:19 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 11:19:19 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
	<5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
Message-ID: <6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>

Well its a standard SIP Refer, They may not support it for good reason.

/b

On Dec 18, 2008, at 11:07 AM, Dennis wrote:

> is deflect, what i understand? the provider has to support it? if yes,
> what could i tell and ask the provider, to find a solution to this
> problem? the provider is quite open for new ideas, although we do not
> want to be to dependant on the provider and his possibilities.




From anthony.minessale at gmail.com  Thu Dec 18 09:20:57 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Dec 2008 11:20:57 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
Message-ID: <191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>

It seems to be related to 20ms vs 30ms ptime.

What are the 2 devices and what rev of FS are you on?

There was more code added in the last few weeks to smooth out this
occurrence.

you can also opt to declare your codec prefs at 30ms

PCMU at 30i instead of just PCMU in the conf.



On Thu, Dec 18, 2008 at 11:05 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello
>
> I got problems with choppy voice. I just happens1 time of 10 or something
> like that.
> Incorrect call: http://pastebin.freeswitch.org/6479
> working call: http://pastebin.freeswitch.org/6481
>
> Any idea what is wrong?
>
> Thanks,
>   Jonas
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Thu Dec 18 09:27:15 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 18:27:15 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
	<5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
	<6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>
Message-ID: <5e414ed0812180927l1ec186efk8c8bdb74337896a@mail.gmail.com>

so if they do not suport it (which has to be seen), is there another
way to redirect a call from one fs to another without the provider?
like redirect from one fs to the other over the local lan?



2008/12/18 Brian West <brian at freeswitch.org>:
> Well its a standard SIP Refer, They may not support it for good reason.
>
> /b
>
> On Dec 18, 2008, at 11:07 AM, Dennis wrote:
>
>> is deflect, what i understand? the provider has to support it? if yes,
>> what could i tell and ask the provider, to find a solution to this
>> problem? the provider is quite open for new ideas, although we do not
>> want to be to dependant on the provider and his possibilities.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Thu Dec 18 09:29:19 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 11:29:19 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180927l1ec186efk8c8bdb74337896a@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
	<5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
	<6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>
	<5e414ed0812180927l1ec186efk8c8bdb74337896a@mail.gmail.com>
Message-ID: <81964BF7-5EDA-47AC-A1E5-81D1046AD977@freeswitch.org>

do they follow a 302 redirect?  Because if the call isn't answered yet  
then you can do a redirect

/b

On Dec 18, 2008, at 11:27 AM, Dennis wrote:

> so if they do not suport it (which has to be seen), is there another
> way to redirect a call from one fs to another without the provider?
> like redirect from one fs to the other over the local lan?




From jonas.gauffin at gmail.com  Thu Dec 18 09:30:14 2008
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 18 Dec 2008 18:30:14 +0100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
Message-ID: <d021d3be0812180930p5edcffewc828110d8dced7ea@mail.gmail.com>

Hello
I've checked out the latest trunk, the problem is still left.
Im using a  Linksys SPA8000 analogue telephone adapter as one device. The
other call comes through the sip gateway (from PSTN).

I'll try to specify 30ms.

Regards,
  Jonas

On Thu, Dec 18, 2008 at 6:20 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> It seems to be related to 20ms vs 30ms ptime.
>
> What are the 2 devices and what rev of FS are you on?
>
> There was more code added in the last few weeks to smooth out this
> occurrence.
>
> you can also opt to declare your codec prefs at 30ms
>
> PCMU at 30i instead of just PCMU in the conf.
>
>
>
> On Thu, Dec 18, 2008 at 11:05 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>
>> Hello
>>
>> I got problems with choppy voice. I just happens1 time of 10 or something
>> like that.
>> Incorrect call: http://pastebin.freeswitch.org/6479
>> working call: http://pastebin.freeswitch.org/6481
>>
>> Any idea what is wrong?
>>
>> Thanks,
>>   Jonas
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Thu Dec 18 09:32:47 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Dec 2008 18:32:47 +0100
Subject: [Freeswitch-users] mod_shout and mp3 formats
In-Reply-To: <49475FDE.7080108@gmx.net>
References: <49475FDE.7080108@gmx.net>
Message-ID: <494A893F.9080509@gmx.net>

Today I also tried playing a wav file with the "play" application and it
worked. However accessing the same file through shout:// didn't work
with freeswitch (with Totem it worked).
The point is that FS plays the file for several seconds, but I don't
hear any sound.
I also looked at the libraries according to the wiki wiich should be
iunstalled, and they are there: Configure does not show any warnings.

Nobody has a clue what may be the problem here?

Best regards
Peter



Peter P GMX schrieb:
> I try to play mp3 I generated through Cepstral TTs and which I encoded
> via lame.
> However they won't play, so my question is: Which mp3 formats are supported?
>
> I generate the wav files by the following
> /opt/swift/bin/swift -n Katrin -p
> audio/channels=1,cst/f0_shift=.8,speech/rate=120,audio/sampling-rate=8000,audio/deadair=2
> -o $wavefile $text
>
> Then I convert to mp3 by the following variations:
> lame 46.wav 46.mp3
> lame -s 32 46.wav 46.mp3
> lame --preset 128 46.wav 46.mp3
> lame --resample 44.1 --preset 128 46.wav 46.mp3
> lame --resample 32 --preset 128 46.wav 46.mp3
> lame --resample 44.1 46.wav 46.mp3
> lame --resample 44.1 -m s --preset 128 46.wav 46.mp3
> lame --resample 44.1 -m s  46.wav 46.mp3
> lame --resample 44.1 -m s -b 128 46.wav 46.mp3
> lame --resample 44.1 -m s -B 24 46.wav 46.mp3
> lame --preset voice -v -B 64 -a 46.wav 46.mp3
>
> None of them worked with the playback application
> (shout://localhost/tts/46.mp3). The sound files had a length of between
> 2 and 5 sec. 2 Times during various tries they played at least
> partially. But at the next try they didn't play again. However I have a
> prerecorded sound file (44.1KHz, 128 kBits stereo music) which always
> plays well.
> The console shows me that all files are successfully played and I get a
> channel_ececute and a channel_ececute_complete after some seconds during
> event_socket. But I don't hear any sound.
>
> All above samples however played well with Totem on Ubuntu.
>
> The wiki tells me that almost any mp3 format should play. What am I
> doing wrong here?
>
> Another question: Should normal wav files play as well? Also with wav I
> cannot hear any sound.
>
> Best regards
> Peter
>
>
>
>
>
>   



From odermann at googlemail.com  Thu Dec 18 09:41:47 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 18:41:47 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <81964BF7-5EDA-47AC-A1E5-81D1046AD977@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<283B4DFE-E5D4-4AC3-9724-BF6E8CF62BD8@freeswitch.org>
	<5e414ed0812180852y35ff0679la26ef24f3140f25d@mail.gmail.com>
	<704A22B0-8410-4D76-A365-948870730FDE@freeswitch.org>
	<5e414ed0812180907u7dca42abu34d223e39e18613c@mail.gmail.com>
	<6F4212D5-01FC-4AFE-9FF6-2AC80E9D59F1@freeswitch.org>
	<5e414ed0812180927l1ec186efk8c8bdb74337896a@mail.gmail.com>
	<81964BF7-5EDA-47AC-A1E5-81D1046AD977@freeswitch.org>
Message-ID: <5e414ed0812180941o4699286hc500a9d8d8d67ea9@mail.gmail.com>

sorry, this is to difficult for me. what does that mean?

they pass a call to one of our fs. then we see, that the call should
be on another fs. we know, that the call is on the wrong fs, before we
send an answer. so we could react accordingly.



2008/12/18 Brian West <brian at freeswitch.org>:
> do they follow a 302 redirect?  Because if the call isn't answered yet
> then you can do a redirect



From gkuri at ieee.org  Thu Dec 18 09:43:57 2008
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 18 Dec 2008 09:43:57 -0800
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
Message-ID: <494A8BDD.1070007@ieee.org>

I've tried to do the same and in my own experience, most carriers don't
accept 302 redirects. What I've seen is they take the 302 as a failure
and move on to the next switch, so worse case with 3 switches, it will
take 2 retries before hitting the switch you want them to redirect to.

Gabe

Dennis wrote:
> i would like to know, what the best way is, to redirect an incoming
> call from one fs (fs1) to another fs (fs2).
> 
> we use 3 freeswitch servers and the carrier passes calls to the three
> fs servers randomly. if on fs server is not offline, the carrier sends
> the call to the next fs.
> this is generally good, but for conferencing it not so good.
> 
> i am using socket outbound and need to do this for conferencing. let's
> say, we have a conference going on on fs1. another person wants to
> enter this conference, but the call is passed to fs2. on fs2 we see,
> that the caller wants to enter the conference going on on fs1.
> 
> now we have to redirect the call from fs2 to fs1. is this done with
> "redirect" and some according settings/params or are there other ways
> to do this? we would like to do this without our carrier doing
> something, to be a little more independant.
> 
> thanks
> dennis
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mszlazak at aol.com  Thu Dec 18 09:45:24 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Thu, 18 Dec 2008 12:45:24 -0500
Subject: [Freeswitch-users] Crackling noise when bypassing media between
	endpoints.
Message-ID: <8CB2F4F8A23FF73-CC8-4D0@Webmail-mg18.sim.aol.com>

When using bypass_media (aka. no_media) mode between an X-lite softphone and Prophacy ASR, I get intermittent "crackiling" background noise with the audio that I'm hearing. 
How do I get rid of this?
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From odermann at googlemail.com  Thu Dec 18 09:58:21 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 18:58:21 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <494A8BDD.1070007@ieee.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org>
Message-ID: <5e414ed0812180958x23fc0836v89c9ddab742e6895@mail.gmail.com>

so at least they should react on a 302? this could help, although i do
not really understand, what happens on a 302.

if they support it, they would receive the target fs server ip where
they should try next with deflect?

if everything does not help and is not possible: what could i do else?
it would be very helpful, if fs would support another way, if the
provider does not offer specific features.



2008/12/18 Gabriel Kuri <gkuri at ieee.org>:
> I've tried to do the same and in my own experience, most carriers don't
> accept 302 redirects. What I've seen is they take the 302 as a failure
> and move on to the next switch, so worse case with 3 switches, it will
> take 2 retries before hitting the switch you want them to redirect to.
>
> Gabe
>
> Dennis wrote:
>> i would like to know, what the best way is, to redirect an incoming
>> call from one fs (fs1) to another fs (fs2).
>>
>> we use 3 freeswitch servers and the carrier passes calls to the three
>> fs servers randomly. if on fs server is not offline, the carrier sends
>> the call to the next fs.
>> this is generally good, but for conferencing it not so good.
>>
>> i am using socket outbound and need to do this for conferencing. let's
>> say, we have a conference going on on fs1. another person wants to
>> enter this conference, but the call is passed to fs2. on fs2 we see,
>> that the caller wants to enter the conference going on on fs1.
>>
>> now we have to redirect the call from fs2 to fs1. is this done with
>> "redirect" and some according settings/params or are there other ways
>> to do this? we would like to do this without our carrier doing
>> something, to be a little more independant.
>>
>> thanks
>> dennis
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From intralanman at freeswitch.org  Thu Dec 18 09:58:41 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 18 Dec 2008 12:58:41 -0500
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <494A8BDD.1070007@ieee.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org>
Message-ID: <494A8F51.3040003@freeswitch.org>

Gabriel Kuri wrote:
> I've tried to do the same and in my own experience, most carriers don't
> accept 302 redirects. What I've seen is they take the 302 as a failure
> and move on to the next switch, so worse case with 3 switches, it will
> take 2 retries before hitting the switch you want them to redirect to.
>
>   

could also just respond with a 503 in which case all carriers should 
fail over to the next one...

-Ray



From sicfslist at gmail.com  Thu Dec 18 10:04:08 2008
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Thu, 18 Dec 2008 12:04:08 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <494A8F51.3040003@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
Message-ID: <35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>

I agree with Ray ... using a 3XX series message is a bad idea ... or you
could put OpenSer in front using the LCR module ... 503 to OpenSer and it
would route to the next gateway in the gateway group.
I have yet to work with any carrier that handles 3XX series correctly except
for some of the TCAP guys.

On Thu, Dec 18, 2008 at 11:58 AM, Raymond Chandler <
intralanman at freeswitch.org> wrote:

> Gabriel Kuri wrote:
> > I've tried to do the same and in my own experience, most carriers don't
> > accept 302 redirects. What I've seen is they take the 302 as a failure
> > and move on to the next switch, so worse case with 3 switches, it will
> > take 2 retries before hitting the switch you want them to redirect to.
> >
> >
>
> could also just respond with a 503 in which case all carriers should
> fail over to the next one...
>
> -Ray
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From c_cav_01 at yahoo.com  Thu Dec 18 10:14:56 2008
From: c_cav_01 at yahoo.com (Chris)
Date: Thu, 18 Dec 2008 10:14:56 -0800 (PST)
Subject: [Freeswitch-users] Crackling noise when bypassing media between
	endpoints.
In-Reply-To: <8CB2F4F8A23FF73-CC8-4D0@Webmail-mg18.sim.aol.com>
Message-ID: <690698.862.qm@web55103.mail.re4.yahoo.com>

I'm no expert, but I believe in media bypass mode freeswitch isn't handling media so it's not a fs fix, it would be the quality of connection for each of the originator/terminator, fs just directs each endpoint to set's up a point to point connection for RTP.

Is this right?

mszlazak at aol.com wrote: When using bypass_media (aka. no_media) mode between an X-lite softphone and Prophacy ASR, I get intermittent "crackiling" background noise with the audio that I'm hearing. 
 How do I get rid of this?
 

---------------------------------
Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 _______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From c_cav_01 at yahoo.com  Thu Dec 18 10:18:36 2008
From: c_cav_01 at yahoo.com (Chris)
Date: Thu, 18 Dec 2008 10:18:36 -0800 (PST)
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <5e414ed0812180958x23fc0836v89c9ddab742e6895@mail.gmail.com>
Message-ID: <476921.94215.qm@web55103.mail.re4.yahoo.com>

Could you set up the 3 inbound call handlers, then set up a 4th switch with the "conference" domain, or if you don't want to set up a 4th, designate one of the 3 inbound switches with a "conference" domain to handle all conferences, do media bypass and bridge the calls to the "conference" domain?

Dennis <odermann at googlemail.com> wrote: so at least they should react on a 302? this could help, although i do
not really understand, what happens on a 302.

if they support it, they would receive the target fs server ip where
they should try next with deflect?

if everything does not help and is not possible: what could i do else?
it would be very helpful, if fs would support another way, if the
provider does not offer specific features.



2008/12/18 Gabriel Kuri :
> I've tried to do the same and in my own experience, most carriers don't
> accept 302 redirects. What I've seen is they take the 302 as a failure
> and move on to the next switch, so worse case with 3 switches, it will
> take 2 retries before hitting the switch you want them to redirect to.
>
> Gabe
>
> Dennis wrote:
>> i would like to know, what the best way is, to redirect an incoming
>> call from one fs (fs1) to another fs (fs2).
>>
>> we use 3 freeswitch servers and the carrier passes calls to the three
>> fs servers randomly. if on fs server is not offline, the carrier sends
>> the call to the next fs.
>> this is generally good, but for conferencing it not so good.
>>
>> i am using socket outbound and need to do this for conferencing. let's
>> say, we have a conference going on on fs1. another person wants to
>> enter this conference, but the call is passed to fs2. on fs2 we see,
>> that the caller wants to enter the conference going on on fs1.
>>
>> now we have to redirect the call from fs2 to fs1. is this done with
>> "redirect" and some according settings/params or are there other ways
>> to do this? we would like to do this without our carrier doing
>> something, to be a little more independant.
>>
>> thanks
>> dennis
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
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Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mike at jerris.com  Thu Dec 18 10:21:18 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Dec 2008 13:21:18 -0500
Subject: [Freeswitch-users] Gtalk to sip problems when reconfiguring
	from scratch
In-Reply-To: <op.umc43xs3atcbto@linux.site>
References: <op.umc43xs3atcbto@linux.site>
Message-ID: <E2C95942-F4AA-4AE6-8E97-DAFADE7967DD@jerris.com>

We do not support registration fetching.

Mike

On Dec 18, 2008, at 9:56 AM, kriko wrote:

> I recently purged all freeswitch config and restarted configuring from
> scratch. Using defaults, I modified public.xml dialplan config  
> (added line
> 16 - 28):
> http://pastebin.com/m5ece6e6f
>
> and added a new config under jingle_profiles:
> http://pastebin.com/d6e983b99
>
> I register with phonelite or twinkle as user 1000 (it says successfull
> reg.), but when I make a call from gtalk I hear that user is not
> available, none of
> client rings.
>
> When registering sofia prints:
> 2008-12-18 15:53:32 [INFO] sofia_presence.c:475
> actual_sofia_presence_event_handler() internal  
> START_PRESENCE_PROBE_SQL
> 2008-12-18 15:53:32 [NOTICE] sofia_presence.c:793
> sofia_presence_resub_callback() internal PRESENCE_PROBE 1000 at 10.99.8.221
> 2008-12-18 15:53:32 [INFO] sofia_presence.c:484
> actual_sofia_presence_event_handler() internal END_PRESENCE_PROBE_SQL
>
> 2008-12-18 15:53:32 [INFO] sofia_presence.c:547
> actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal)
> 2008-12-18 15:53:32 [INFO] sofia_presence.c:563
> actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal)
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:517
> actual_sofia_presence_event_handler() external is passive, skipping
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510
> actual_sofia_presence_event_handler() 10.99.8.221 is an alias,  
> skipping
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510
> actual_sofia_presence_event_handler() default is an alias, skipping
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510
> actual_sofia_presence_event_handler() nat is an alias, skipping
> 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510
> actual_sofia_presence_event_handler() outbound is an alias, skipping
>
> and when fetching registrations:
> 2008-12-18 15:53:50 [ERR] sofia_reg.c:1120
> sofia_reg_handle_sip_i_register() NO CONTACT!
>
> Does this means that registration failed? Why it doesn't call  
> anymore the
> targeted user (1000)?
>
> Here is also an extract from console log while calling, it caught my
> attention:
> 2008-12-18 15:54:41 [WARNING] mod_dptools.c:2047  
> user_outgoing_channel()
> Can't find user [1000@]
> 2008-12-18 15:54:41 [ERR] switch_ivr_originate.c:1110
> switch_ivr_originate() Cannot create outgoing channel of type [user]
> cause: [SUBSCRIBER_ABSENT]
> 2008-12-18 15:54:41 [DEBUG] switch_ivr_originate.c:1689
> switch_ivr_originate() Originate Resulted in Error Cause: 20
> [SUBSCRIBER_ABSENT]
>



From c_cav_01 at yahoo.com  Thu Dec 18 10:28:00 2008
From: c_cav_01 at yahoo.com (Chris)
Date: Thu, 18 Dec 2008 10:28:00 -0800 (PST)
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <476921.94215.qm@web55103.mail.re4.yahoo.com>
Message-ID: <365310.11415.qm@web55103.mail.re4.yahoo.com>

If you need to do load balancing, you could set up a conference_a domain on one switch, conference_b on the second, conference_c on the third, then use xml_curl to dialplan and bridge the call to the right domain...

But again, I am no expert...  Just a noob trying to be creative.  :P

Chris <c_cav_01 at yahoo.com> wrote: Could you set up the 3 inbound call handlers, then set up a 4th switch with the "conference" domain, or if you don't want to set up a 4th, designate one of the 3 inbound switches with a "conference" domain to handle all conferences, do media bypass and bridge the calls to the "conference" domain?

Dennis <odermann at googlemail.com> wrote: so at least they should react on a 302? this could help, although i do
not really understand, what happens on a 302.

if they support it, they would receive the target fs server ip where
they should try next with deflect?

if everything does not help and is not possible: what could i do else?
it would be very helpful, if fs would support another way, if the
provider does not offer specific features.



2008/12/18 Gabriel Kuri :
> I've tried to do  the same and in my own experience, most carriers don't
> accept 302 redirects. What I've seen is they take the 302 as a failure
> and move on to the next switch, so worse case with 3 switches, it will
> take 2 retries before hitting the switch you want them to redirect to.
>
> Gabe
>
> Dennis wrote:
>> i would like to know, what the best way is, to redirect an incoming
>> call from one fs (fs1) to another fs (fs2).
>>
>> we use 3 freeswitch servers and the carrier passes calls to the three
>> fs servers randomly. if on fs server is not offline, the carrier sends
>> the call to the next fs.
>> this is generally good, but for conferencing it not so good.
>>
>> i am using socket outbound and need to do this for conferencing. let's
>> say, we have a conference going on on fs1. another person wants to
>> enter this conference, but the call  is passed to fs2. on fs2 we see,
>> that the caller wants to enter the conference going on on fs1.
>>
>> now we have to redirect the call from fs2 to fs1. is this done with
>> "redirect" and some according settings/params or are there other ways
>> to do this? we would like to do this without our carrier doing
>> something, to be a little more independant.
>>
>> thanks
>> dennis
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
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Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From Prometheus001 at gmx.net  Thu Dec 18 10:29:11 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 18 Dec 2008 19:29:11 +0100
Subject: [Freeswitch-users] mod_shout and mp3 formats
In-Reply-To: <49475FDE.7080108@gmx.net>
References: <49475FDE.7080108@gmx.net>
Message-ID: <494A9677.8030608@gmx.net>

Today I also tried playing a wav file with the "play" application and it
worked. However accessing the same file through shout:// didn't work
with freeswitch (with Totem it worked).
The point is that FS plays the file for several seconds, but I don't
hear any sound.
I also looked at the libraries according to the wiki wiich should be
iunstalled, and they are there: Configure does not show any warnings.

Nobody has a clue what may be the problem here?

Best regards
Peter



Peter P GMX schrieb:
> I try to play mp3 I generated through Cepstral TTs and which I encoded
> via lame.
> However they won't play, so my question is: Which mp3 formats are supported?
>
> I generate the wav files by the following
> /opt/swift/bin/swift -n Katrin -p
> audio/channels=1,cst/f0_shift=.8,speech/rate=120,audio/sampling-rate=8000,audio/deadair=2
> -o $wavefile $text
>
> Then I convert to mp3 by the following variations:
> lame 46.wav 46.mp3
> lame -s 32 46.wav 46.mp3
> lame --preset 128 46.wav 46.mp3
> lame --resample 44.1 --preset 128 46.wav 46.mp3
> lame --resample 32 --preset 128 46.wav 46.mp3
> lame --resample 44.1 46.wav 46.mp3
> lame --resample 44.1 -m s --preset 128 46.wav 46.mp3
> lame --resample 44.1 -m s  46.wav 46.mp3
> lame --resample 44.1 -m s -b 128 46.wav 46.mp3
> lame --resample 44.1 -m s -B 24 46.wav 46.mp3
> lame --preset voice -v -B 64 -a 46.wav 46.mp3
>
> None of them worked with the playback application
> (shout://localhost/tts/46.mp3). The sound files had a length of between
> 2 and 5 sec. 2 Times during various tries they played at least
> partially. But at the next try they didn't play again. However I have a
> prerecorded sound file (44.1KHz, 128 kBits stereo music) which always
> plays well.
> The console shows me that all files are successfully played and I get a
> channel_ececute and a channel_ececute_complete after some seconds during
> event_socket. But I don't hear any sound.
>
> All above samples however played well with Totem on Ubuntu.
>
> The wiki tells me that almost any mp3 format should play. What am I
> doing wrong here?
>
> Another question: Should normal wav files play as well? Also with wav I
> cannot hear any sound.
>
> Best regards
> Peter
>
>
>
>
>
>   




From brian at freeswitch.org  Thu Dec 18 10:31:11 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 12:31:11 -0600
Subject: [Freeswitch-users] Crackling noise when bypassing media between
	endpoints.
In-Reply-To: <690698.862.qm@web55103.mail.re4.yahoo.com>
References: <690698.862.qm@web55103.mail.re4.yahoo.com>
Message-ID: <22787EB4-371A-4907-8C9D-96D7F2782AF2@freeswitch.org>

Yes Chris you are right.  FreeSWITCH isn't involved in the media at all.

/b

On Dec 18, 2008, at 12:14 PM, Chris wrote:

> I'm no expert, but I believe in media bypass mode freeswitch isn't  
> handling media so it's not a fs fix, it would be the quality of  
> connection for each of the originator/terminator, fs just directs  
> each endpoint to set's up a point to point connection for RTP.
>
> Is this right?




From odermann at googlemail.com  Thu Dec 18 10:36:04 2008
From: odermann at googlemail.com (Dennis)
Date: Thu, 18 Dec 2008 19:36:04 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <494A8F51.3040003@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
Message-ID: <5e414ed0812181036q297d7f82vcec9b8b0e3cf8cee@mail.gmail.com>

thanks for all your help!

this sounds interesting. it seems, that these codes should be
available by default with sip!? is this right?

i will talk to the carrier tomorrow and ask, what is possible.

as far as i can see, i am always dependant on the carrier? there is no
way to pass a call from one fs to another?




2008/12/18 Raymond Chandler <intralanman at freeswitch.org>:
> Gabriel Kuri wrote:
>> I've tried to do the same and in my own experience, most carriers don't
>> accept 302 redirects. What I've seen is they take the 302 as a failure
>> and move on to the next switch, so worse case with 3 switches, it will
>> take 2 retries before hitting the switch you want them to redirect to.
>>
>>
>
> could also just respond with a 503 in which case all carriers should
> fail over to the next one...
>
> -Ray
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Thu Dec 18 10:40:03 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 18 Dec 2008 13:40:03 -0500
Subject: [Freeswitch-users] mod_shout and mp3 formats
In-Reply-To: <494A9677.8030608@gmx.net>
References: <49475FDE.7080108@gmx.net> <494A9677.8030608@gmx.net>
Message-ID: <04209D71-419B-4346-9A9F-C9D85B566A8F@jerris.com>

shout does not play wav files it plays mp3 files.

Mike

On Dec 18, 2008, at 1:29 PM, Peter P GMX wrote:

> Today I also tried playing a wav file with the "play" application  
> and it
> worked. However accessing the same file through shout:// didn't work
> with freeswitch (with Totem it worked).
> The point is that FS plays the file for several seconds, but I don't
> hear any sound.
> I also looked at the libraries according to the wiki wiich should be
> iunstalled, and they are there: Configure does not show any warnings.
>
> Nobody has a clue what may be the problem here?
>
> Best regards
> Peter




From mszlazak at aol.com  Thu Dec 18 10:49:02 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Thu, 18 Dec 2008 13:49:02 -0500
Subject: [Freeswitch-users] Crackling noise when bypassing media between
 endpoints.
In-Reply-To: <22787EB4-371A-4907-8C9D-96D7F2782AF2@freeswitch.org>
References: <690698.862.qm@web55103.mail.re4.yahoo.com>
	<22787EB4-371A-4907-8C9D-96D7F2782AF2@freeswitch.org>
Message-ID: <8CB2F586D303ED3-CC8-8B6@Webmail-mg18.sim.aol.com>


 Man, I can't win with this one. 

I can bypass media between two endpoints with some "static" but what I really want FS to do is process the audio before it's passed on.
However, getting FS involved is something I haven't had any success in with these two endpoints ... so far.

Thanks for pointing out the noise source(s) with bypass ... makes sense given the name.


 


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, 18 Dec 2008 10:31 am
Subject: Re: [Freeswitch-users] Crackling noise when bypassing media between endpoints.










Yes Chris you are right.  FreeSWITCH isn't involved in the media at all.

/b

On Dec 18, 2008, at 12:14 PM, Chris wrote:

> I'm no expert, but I believe in media bypass mode freeswitch isn't  
> handling media so it's not a fs fix, it would be the quality of  
> connection for each of the originator/terminator, fs just directs  
> each endpoint to set's up a point to point connection for RTP.
>
> Is this right?


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From anthony.minessale at gmail.com  Thu Dec 18 11:00:26 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Dec 2008 13:00:26 -0600
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <494A6DE0.4030701@networkoblivion.com>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<494A6DE0.4030701@networkoblivion.com>
Message-ID: <191c3a030812181100jd7a0c20m97596121650d404b@mail.gmail.com>

Can you answer the questions and possibly go online on IRC so we can debug
your issue?


On Thu, Dec 18, 2008 at 9:36 AM, peder at networkoblivion.com <
peder at networkoblivion.com> wrote:

> If anybody wants to look at the core dump in gdb, here it is (the actual
> core is 256Meg):
>
> http://pastebin.freeswitch.org/6476
>
> I know zip about debugging and gdb, but from looking through it, I see a
> segmentation fault and it appears to be thread 15094.  The last three
> items in the bt full for that thread are:
>
> destroy_status = <value optimized out>
> fd = (switch_file_t *) 0x80529b0
> pool = (switch_memory_pool_t *) 0x80528f0
>
> So I would guess it is trying to access an invalid memory location, but
> why, I have no idea....
>
> Any ideas?
>
>
> Peder
>
>
> Michael S Collins wrote:
> > Check out this page:
> > wiki.freeswitch.org/wiki/Debugging_Freeswitch
> >
> > -MC
> >
> > Sent from my iPhone
> >
> > On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <
> peder at networkoblivion.com
> >  > wrote:
> >
> >> What is the process for capturing and submitting a core dump?
> >>
> >> I am messing around with the Cisco 79x1 phones and tcp and multiple
> >> reg.
> >> I have a 7961 using tcp and a 7960 using udp both reg'd with the same
> >> number and both showing up as registered.  If I call out from the
> >> phone
> >> using tcp, it works.  If I call out from the phone using udp, I get a
> >> core dump.  If I call in, it calls both phones and I am assuming the
> >> call to the phone using udp causes a core dump as well.  These are the
> >> only two phones on the system and I am running version 10851 from
> >> yesterday.  If I only have the udp phone registered and the tcp
> >> phone is
> >> off, it works fine.  It is only when I have a mix of a udp reg and a
> >> tcp
> >> reg on the same number that I appear to get a core dump.
> >>
> >>
> >> Peder
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From intralanman at freeswitch.org  Thu Dec 18 11:05:06 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 18 Dec 2008 14:05:06 -0500
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <21069519.post@talk.nabble.com>
References: <21053181.post@talk.nabble.com>
	<49490301.8040701@lists.rupa.com>	<21054080.post@talk.nabble.com>
	<49491847.3030403@lists.rupa.com> <21069519.post@talk.nabble.com>
Message-ID: <494A9EE2.7050507@freeswitch.org>

Carole O. wrote:
> Hello,
>
> Thanks for your answers!
> Concerning the creation of a new variable for the conference the problem is
> that I do not create channels from the conference. I call separately a new
> member on a new channel and add it on the conference only if he agrees to
> enter it. So it was the same problem as for the uuid, I am not sure I can
> access the good variable from anywhere in case many conferences are running.
>
>   
you could use the db app to hold state across multiple calls... maybe 
use the ${caller_id_number} and the ${destination_number} as keys for 
the insert/select so that there's something constant to use in the 
select... and another extension or two may be needed...

You could do the db lookup before you make the call so that you see if 
your caller is already a member of a conference.... if he is, then the 
transfer from *1 would work much the same as it does now except you'd 
use the result of the db lookup as the conference number... if he's not 
a member of an existing conference, then you could generate the uuid 
like Anthony said before, then do a db insert for ${caller_id_number} 
and ${destination_number} to insert that newly created uuid and use it 
as the conference number.... one caveat that i see here is that the 
destination_number would have to be exactly the same as if that user 
were callling and it was his caller_id_number, otherwise your query will 
fail.

you'll also need to "clean" the db when you hangup, which should be able 
to be accomplished with an execute_on_hangup that does a delete of the 
conf data for each user

-Ray



From kkielhofner at star2star.com  Thu Dec 18 11:21:49 2008
From: kkielhofner at star2star.com (Kristian Kielhofner)
Date: Thu, 18 Dec 2008 14:21:49 -0500
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
	<35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>
Message-ID: <2d9149cd0812181121v47823968u6fb8f40aa86c5fb3@mail.gmail.com>

On Thu, Dec 18, 2008 at 1:04 PM, Shelby Ramsey <sicfslist at gmail.com> wrote:
> I agree with Ray ... using a 3XX series message is a bad idea ... or you
> could put OpenSer in front using the LCR module ... 503 to OpenSer and it
> would route to the next gateway in the gateway group.
> I have yet to work with any carrier that handles 3XX series correctly except
> for some of the TCAP guys.
>

Level(3) readily supports 302s if the destination IP of the new
contact has been made known to Level(3) beforehand.  You can't 302
just anywhere but you can 302 to your own boxes, networks, etc.



-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From brian at freeswitch.org  Thu Dec 18 11:27:53 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Dec 2008 13:27:53 -0600
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <2d9149cd0812181121v47823968u6fb8f40aa86c5fb3@mail.gmail.com>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
	<35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>
	<2d9149cd0812181121v47823968u6fb8f40aa86c5fb3@mail.gmail.com>
Message-ID: <A8991787-3B4E-480C-A137-685C5884BD8E@freeswitch.org>

Excellent advice.   So just letting L3 know the IP's and you'll be fine.

/b

On Dec 18, 2008, at 1:21 PM, Kristian Kielhofner wrote:

>
> Level(3) readily supports 302s if the destination IP of the new
> contact has been made known to Level(3) beforehand.  You can't 302
> just anywhere but you can 302 to your own boxes, networks, etc.




From peder at networkoblivion.com  Thu Dec 18 11:33:31 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Thu, 18 Dec 2008 13:33:31 -0600
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <CAEDC248-D670-4603-A4A8-5F06074304E9@freeswitch.org>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>	<494A6DE0.4030701@networkoblivion.com>
	<CAEDC248-D670-4603-A4A8-5F06074304E9@freeswitch.org>
Message-ID: <494AA58B.9030500@networkoblivion.com>

I can make it happen on demand.  All I have to do is call the shared 
number and it crashes.  I'll hop on IRC in a bit.

Michael S Collins wrote:
> Is this a single occurrence or can you make it happen consistently?
> -MC
> 
> Sent from my iPhone
> 
> On Dec 18, 2008, at 7:36 AM, "peder at networkoblivion.com" <peder at networkoblivion.com 
>  > wrote:
> 
>> If anybody wants to look at the core dump in gdb, here it is (the  
>> actual
>> core is 256Meg):
>>
>> http://pastebin.freeswitch.org/6476
>>
>> I know zip about debugging and gdb, but from looking through it, I  
>> see a
>> segmentation fault and it appears to be thread 15094.  The last three
>> items in the bt full for that thread are:
>>
>> destroy_status = <value optimized out>
>> fd = (switch_file_t *) 0x80529b0
>> pool = (switch_memory_pool_t *) 0x80528f0
>>
>> So I would guess it is trying to access an invalid memory location,  
>> but
>> why, I have no idea....
>>
>> Any ideas?
>>
>>
>> Peder
>>
>>
>> Michael S Collins wrote:
>>> Check out this page:
>>> wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>
>>> -MC
>>>
>>> Sent from my iPhone
>>>
>>> On Dec 18, 2008, at 6:38 AM, "peder at networkoblivion.com" <peder at networkoblivion.com
>>>> wrote:
>>>> What is the process for capturing and submitting a core dump?
>>>>
>>>> I am messing around with the Cisco 79x1 phones and tcp and multiple
>>>> reg.
>>>> I have a 7961 using tcp and a 7960 using udp both reg'd with the  
>>>> same
>>>> number and both showing up as registered.  If I call out from the
>>>> phone
>>>> using tcp, it works.  If I call out from the phone using udp, I  
>>>> get a
>>>> core dump.  If I call in, it calls both phones and I am assuming the
>>>> call to the phone using udp causes a core dump as well.  These are  
>>>> the
>>>> only two phones on the system and I am running version 10851 from
>>>> yesterday.  If I only have the udp phone registered and the tcp
>>>> phone is
>>>> off, it works fine.  It is only when I have a mix of a udp reg and a
>>>> tcp
>>>> reg on the same number that I appear to get a core dump.
>>>>
>>>>
>>>> Peder
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From scott.ellis at novatex.com.au  Thu Dec 18 15:04:54 2008
From: scott.ellis at novatex.com.au (Scott Ellis)
Date: Fri, 19 Dec 2008 10:04:54 +1100
Subject: [Freeswitch-users] Pennytel Gateway Registration problem
In-Reply-To: <191c3a030812180612n94f9b72vc9ad0ba2d90d6a9e@mail.gmail.com>
References: <49478B0F.3000802@novatex.com.au>	<153EBC09-A97F-4806-9EF7-2FB6DEC46E70@jerris.com>	<4949B946.5050502@novatex.com.au>
	<191c3a030812180612n94f9b72vc9ad0ba2d90d6a9e@mail.gmail.com>
Message-ID: <494AD716.1010708@novatex.com.au>

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From jason at jasonjgw.net  Thu Dec 18 20:35:09 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Dec 2008 15:35:09 +1100
Subject: [Freeswitch-users] debug symbols (was Re: Core Dump)
In-Reply-To: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
Message-ID: <20081219043509.GA4225@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 07:02:18AM -0800, Michael S Collins wrote:
> Check out this page:
> wiki.freeswitch.org/wiki/Debugging_Freeswitch

In the long term (i.e., when more important matters aren't at issue), it might
be a good idea to modify the build process so that the debug symbols are
written out as separate files. In recent Linux distributions such as Debian
and Fedora, the debug symbols are often provided in separate packages that can
be installed whenever needed.




From marc at kasteris.com  Thu Dec 18 21:46:39 2008
From: marc at kasteris.com (Marc Orenberg)
Date: Thu, 18 Dec 2008 21:46:39 -0800 (PST)
Subject: [Freeswitch-users] Ending a bridged call with a touchtone
References: <mailman.2512.1229488183.18013.freeswitch-users@lists.freeswitch.org>
Message-ID: <876517.26535.qm@web50805.mail.re2.yahoo.com>

Thanks for the response Brian.? I don't understand what bind_meta does, or how it can help me. Is it something I can use from my Python script?? I searched for a description of it, but I was unable to find one.? Could you please point me towards some documentation, or maybe quickly explain it?? Thanks.

?
>Date: Tue, 16 Dec 2008 11:30:46 -0600
>From: Brian West <brian at freeswitch.org>
>Subject: Re: [Freeswitch-users] Ending a bridged call with a touchtone
>To: freeswitch-users at lists.freeswitch.org
>Message-ID: <F9E12702-5CA0-45A8-B408-3F2162F94A8E at freeswitch.org>
>Content-Type: text/plain; charset="us-ascii"
>
>Try bind_meta, examples are in the default dialplan.
>
>/b
>
>On Dec 16, 2008, at 10:24 PM, Marc Orenberg wrote:
>
>> Hello.? I'm trying to allow the A-leg of a bridged call to be able? 
>> to press a touchtone to end the call.
>> In my Python script, I set-up a DTMF callback function using? 
>> setInputCallback, but it doesn't seem to have any effect during? 
>> bridged calls. Is there another way to do this? Thanks.
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From mike at jerris.com  Thu Dec 18 21:24:24 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Dec 2008 00:24:24 -0500
Subject: [Freeswitch-users] debug symbols (was Re: Core Dump)
In-Reply-To: <20081219043509.GA4225@jdc.jasonjgw.net>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<20081219043509.GA4225@jdc.jasonjgw.net>
Message-ID: <02B71311-5C31-4B19-BD62-6DBC9958F774@jerris.com>

This is a function of the packaging system, not the build system and  
at least the debs do have this already.  On a related note I fixed the  
Sofia build to include proper symbols now all the time in the debug  
build.

Mike

On Dec 18, 2008, at 11:35 PM, Jason White <jason at jasonjgw.net> wrote:

> On Thu, Dec 18, 2008 at 07:02:18AM -0800, Michael S Collins wrote:
>> Check out this page:
>> wiki.freeswitch.org/wiki/Debugging_Freeswitch
>
> In the long term (i.e., when more important matters aren't at  
> issue), it might
> be a good idea to modify the build process so that the debug symbols  
> are
> written out as separate files. In recent Linux distributions such as  
> Debian
> and Fedora, the debug symbols are often provided in separate  
> packages that can
> be installed whenever needed.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason at jasonjgw.net  Thu Dec 18 23:41:37 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Dec 2008 18:41:37 +1100
Subject: [Freeswitch-users] debug symbols (was Re: Core Dump)
In-Reply-To: <02B71311-5C31-4B19-BD62-6DBC9958F774@jerris.com>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<20081219043509.GA4225@jdc.jasonjgw.net>
	<02B71311-5C31-4B19-BD62-6DBC9958F774@jerris.com>
Message-ID: <20081219074137.GA6310@jdc.jasonjgw.net>

On Fri, Dec 19, 2008 at 12:24:24AM -0500, Michael Jerris wrote:
> This is a function of the packaging system, not the build system and  
> at least the debs do have this already.  On a related note I fixed the  
> Sofia build to include proper symbols now all the time in the debug  
> build.

Thanks, this is excellent.

The supportive community associated with this project is gratefully
appreciated.




From msc at freeswitch.org  Thu Dec 18 23:49:35 2008
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Dec 2008 23:49:35 -0800
Subject: [Freeswitch-users] debug symbols (was Re: Core Dump)
In-Reply-To: <20081219074137.GA6310@jdc.jasonjgw.net>
References: <4B9425AB-E078-4ADB-802F-103488E85747@freeswitch.org>
	<20081219043509.GA4225@jdc.jasonjgw.net>
	<02B71311-5C31-4B19-BD62-6DBC9958F774@jerris.com>
	<20081219074137.GA6310@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812182349kf65c76cr6f010e32e4ed4cd6@mail.gmail.com>

On Thu, Dec 18, 2008 at 11:41 PM, Jason White <jason at jasonjgw.net> wrote:

> On Fri, Dec 19, 2008 at 12:24:24AM -0500, Michael Jerris wrote:
> > This is a function of the packaging system, not the build system and
> > at least the debs do have this already.  On a related note I fixed the
> > Sofia build to include proper symbols now all the time in the debug
> > build.
>
> Thanks, this is excellent.
>
> The supportive community associated with this project is gratefully
> appreciated.


The members of the supportive community appreciate your appreciation! ;)
-MC


>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mszlazak at aol.com  Fri Dec 19 00:03:31 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Fri, 19 Dec 2008 03:03:31 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
Message-ID: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>

I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.
I'm trying to pass audio "internally" on the same machine between endpoints and have be advised that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 
However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.
Any comments?? 
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From fidibus83 at aol.com  Fri Dec 19 01:03:32 2008
From: fidibus83 at aol.com (fidibus83)
Date: Fri, 19 Dec 2008 10:03:32 +0100
Subject: [Freeswitch-users] Problem with openzap
In-Reply-To: <004801c960f1$dc680840$6445310a@Franzi>
References: <004801c960f1$dc680840$6445310a@Franzi>
Message-ID: <005e01c961b8$aba3b840$6445310a@Franzi>

Hello,

 

Here is the newbie in FS! I need your help again!

 

When FS is running I get every few seconds this warning:

 

[WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6 (or 7 or 8)

 

Why?

 

Do you need some configurations?

 

Thanks!

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From kristjan.ugrin at gmail.com  Fri Dec 19 01:09:49 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Fri, 19 Dec 2008 10:09:49 +0100
Subject: [Freeswitch-users] Java example
In-Reply-To: <191c3a030812180619k394a9e33g7cba5808e7d95558@mail.gmail.com>
References: <op.umcs7ncwatcbto@linux.site>
	<6df5b6fa30d3767b81a6a1d9da04d3ac.squirrel@webmail.ecntelecoms.com>
	<op.umcxenzsatcbto@linux.site>
	<191c3a030812180619k394a9e33g7cba5808e7d95558@mail.gmail.com>
Message-ID: <op.umejqlg7atcbto@linux.site>

Seems like my dialplan was a bit problematic, it works now.
Thanks.

On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale  
<anthony.minessale at gmail.com> wrote:

> did you turn up your console log level high enough to see it? The default
> level is "INFO"
>
>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>



-- 
kriko



From kristjan.ugrin at gmail.com  Fri Dec 19 01:12:49 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Fri, 19 Dec 2008 10:12:49 +0100
Subject: [Freeswitch-users] [Java] Catching dingaling messages
Message-ID: <op.umejvnuiatcbto@linux.site>

Hello!

I was wondering if it would be possible to catch messages from dingaling.  
I saw it can print out messages into console when a user types in a  
message,
but it doesn't understand it. I would like to catch that and do something  
(like initiate a call).
I know you have to call you program via a dialplan, so I don't know how  
would it be really possible to invoke your java app
when other events occur.

Is it possible?

Cheers,

-- 
kriko



From jason at jasonjgw.net  Fri Dec 19 01:22:07 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Dec 2008 20:22:07 +1100
Subject: [Freeswitch-users] [Java] Catching dingaling messages
In-Reply-To: <op.umejvnuiatcbto@linux.site>
References: <op.umejvnuiatcbto@linux.site>
Message-ID: <20081219092207.GA6967@jdc.jasonjgw.net>

On Fri, Dec 19, 2008 at 10:12:49AM +0100, kriko wrote:
> I was wondering if it would be possible to catch messages from dingaling.  
> I saw it can print out messages into console when a user types in a  
> message,
> but it doesn't understand it. I would like to catch that and do something  
> (like initiate a call).
> I know you have to call you program via a dialplan, so I don't know how  
> would it be really possible to invoke your java app
> when other events occur.

Listen to the event socket on port 8021 and register to receive the events you
want to monitor, then issue api commands via the same socket interface to make
the outbound call.

See the wiki page about the event socket.

I haven't looked at this in any detail, but it's clear from the wiki that this
should be able to meet your needs.




From jason at jasonjgw.net  Fri Dec 19 02:02:58 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Dec 2008 21:02:58 +1100
Subject: [Freeswitch-users] Ending a bridged call with a touchtone
In-Reply-To: <876517.26535.qm@web50805.mail.re2.yahoo.com>
References: <mailman.2512.1229488183.18013.freeswitch-users@lists.freeswitch.org>
	<876517.26535.qm@web50805.mail.re2.yahoo.com>
Message-ID: <20081219100258.GA7206@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 09:46:39PM -0800, Marc Orenberg wrote:
> Thanks for the response Brian.? I don't understand what bind_meta does, or
> how it can help me. Is it something I can use from my Python script?? I
> searched for a description of it, but I was unable to find one.? Could you
> please point me towards some documentation, or maybe quickly explain it??

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app




From fidibus83 at aol.com  Fri Dec 19 02:10:29 2008
From: fidibus83 at aol.com (fidibus83)
Date: Fri, 19 Dec 2008 11:10:29 +0100
Subject: [Freeswitch-users] Problem with openzap
In-Reply-To: <005e01c961b8$aba3b840$6445310a@Franzi>
References: <004801c960f1$dc680840$6445310a@Franzi>
	<005e01c961b8$aba3b840$6445310a@Franzi>
Message-ID: <006d01c961c2$06085990$6445310a@Franzi>

Hello,

 

I get more warnings yet:

 

[WARNING]  mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for
channel 1:1( it?s going to 1:31)

[WARNING] zap_isdn.c:803 process_event() channel 1:1 (1:2) (it?s going to
1:31 (1:16))

 

I don?t know what to do? Can you help me?

 

I have a Linux-Server with a Digium Wildcard TE110P.

 

Oz list:

 

API CALL [oz(list)] output:

+OK

span: 1

type: isdn

chan_count: 31

Dialplan: XML

context: default

dial_regex:

fial_dial_regex:

hold_music:

analog_options none

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
fidibus83
Gesendet: Freitag, 19. Dezember 2008 10:04
An: freeswitch-users at lists.freeswitch.org
Betreff: [Freeswitch-users] Problem with openzap

 

Hello,

 

Here is the newbie in FS! I need your help again!

 

When FS is running I get every few seconds this warning:

 

[WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6 (or 7 or 8)

 

Why?

 

Do you need some configurations?

 

Thanks!
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From odermann at googlemail.com  Fri Dec 19 04:18:52 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 19 Dec 2008 13:18:52 +0100
Subject: [Freeswitch-users] Redirecting a call from one FS to another FS?
In-Reply-To: <A8991787-3B4E-480C-A137-685C5884BD8E@freeswitch.org>
References: <5e414ed0812180836y530b7250q114fd7ff7068b967@mail.gmail.com>
	<494A8BDD.1070007@ieee.org> <494A8F51.3040003@freeswitch.org>
	<35b355e90812181004ifc9656y8ec493377622780c@mail.gmail.com>
	<2d9149cd0812181121v47823968u6fb8f40aa86c5fb3@mail.gmail.com>
	<A8991787-3B4E-480C-A137-685C5884BD8E@freeswitch.org>
Message-ID: <5e414ed0812190418j19c259e4m674c55fde7765de2@mail.gmail.com>

sendmsg redirect to an ip-adress of one of our fs server works great.
thanks for your help.

dannis



From Claudio.Cavalera at italtel.it  Fri Dec 19 04:36:45 2008
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Fri, 19 Dec 2008 13:36:45 +0100
Subject: [Freeswitch-users] Problem compiling socket2me
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0969642@BESONE.corp.dom>

Hello guys,
I'm playing with fs fax capabilities following these guidelines:
http://wiki.freeswitch.org/wiki/Examples_faxlib.jm

I've compiled mod_fax
with make mod_fax-install
and that should have taken care also of spandsp.

When I issue make in scripts/socket2me I get this error:
socket2me.c:315: error: 'fax_state_t' has no member named 't30_state'

Maybe something has changed in the api?
Could you please help me track down the problem?

Thanks,
Claudio


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From kristjan.ugrin at gmail.com  Fri Dec 19 07:37:52 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Fri, 19 Dec 2008 16:37:52 +0100
Subject: [Freeswitch-users] [Java] Catching dingaling messages
In-Reply-To: <20081219092207.GA6967@jdc.jasonjgw.net>
References: <op.umejvnuiatcbto@linux.site>
	<20081219092207.GA6967@jdc.jasonjgw.net>
Message-ID: <op.ume1pdtjatcbto@linux.site>

That's seems the right this, thanks.

But the dingaling is only returning this events:
  dingaling::login_success
  dingaling::login_failure
  dingaling::connected

Is it possible in any way to catch text messages?


On Fri, 19 Dec 2008 10:22:07 +0100, Jason White <jason at jasonjgw.net> wrote:

> On Fri, Dec 19, 2008 at 10:12:49AM +0100, kriko wrote:
>> I was wondering if it would be possible to catch messages from  
>> dingaling.
>> I saw it can print out messages into console when a user types in a
>> message,
>> but it doesn't understand it. I would like to catch that and do  
>> something
>> (like initiate a call).
>> I know you have to call you program via a dialplan, so I don't know how
>> would it be really possible to invoke your java app
>> when other events occur.
>
> Listen to the event socket on port 8021 and register to receive the  
> events you
> want to monitor, then issue api commands via the same socket interface  
> to make
> the outbound call.
>
> See the wiki page about the event socket.
>
> I haven't looked at this in any detail, but it's clear from the wiki  
> that this
> should be able to meet your needs.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Porn - the reason you need a new hard drive.



From msc at freeswitch.org  Fri Dec 19 07:45:04 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Dec 2008 07:45:04 -0800
Subject: [Freeswitch-users] Problem with openzap
In-Reply-To: <006d01c961c2$06085990$6445310a@Franzi>
References: <004801c960f1$dc680840$6445310a@Franzi>
	<005e01c961b8$aba3b840$6445310a@Franzi>
	<006d01c961c2$06085990$6445310a@Franzi>
Message-ID: <87f2f3b90812190745i2101335fw483990419659603e@mail.gmail.com>

Which dialect are you running and what is on the other end of the PRI?
-MC

On Fri, Dec 19, 2008 at 2:10 AM, fidibus83 <fidibus83 at aol.com> wrote:

>  Hello,
>
>
>
> I get more warnings yet:
>
>
>
> [WARNING]  mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for
> channel 1:1( it's going to 1:31)
>
> [WARNING] zap_isdn.c:803 process_event() channel 1:1 (1:2) (it's going to
> 1:31 (1:16))
>
>
>
> I don't know what to do? Can you help me?
>
>
>
> I have a Linux-Server with a Digium Wildcard TE110P.
>
>
>
> Oz list:
>
>
>
> API CALL [oz(list)] output:
>
> +OK
>
> span: 1
>
> type: isdn
>
> chan_count: 31
>
> Dialplan: XML
>
> context: default
>
> dial_regex:
>
> fial_dial_regex:
>
> hold_music:
>
> analog_options none
>
>
>  ------------------------------
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *fidibus83
> *Gesendet:* Freitag, 19. Dezember 2008 10:04
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* [Freeswitch-users] Problem with openzap
>
>
>
> Hello,
>
>
>
> Here is the newbie in FS! I need your help again!
>
>
>
> When FS is running I get every few seconds this warning:
>
>
>
> [WARNING] zap_zt.c:642 zt_next_event() Unhandled event 6 (or 7 or 8)
>
>
>
> Why?
>
>
>
> Do you need some configurations?
>
>
>
> Thanks!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Fri Dec 19 07:49:33 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Dec 2008 10:49:33 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
Message-ID: <B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>

It gives me the impression there is something wrong with your firewall  
running on the box.

Mike

On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:

> I find it strange that I can have to endpoints get audio went using  
> bypass media mode but the audio fails to go across endpoints if I  
> use proxy media mode.
> I'm trying to pass audio "internally" on the same machine between  
> endpoints and have be advised that a reason the audio may fail to be  
> passed is because there is some RTP timing and IP address/port issues.
> However, FS has no problem "connecting" ports if i change the mode  
> to bypass media. This gives me the impression that something is  
> wrong with FS proxy media mode.
> Any comments?
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Dec 19 07:52:57 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 19 Dec 2008 10:52:57 -0500
Subject: [Freeswitch-users] Problem compiling socket2me
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0969642@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0969642@BESONE.corp.dom>
Message-ID: <94326D53-5AFD-4B71-845D-41F486D02D10@jerris.com>

mod_fax replaces socket2me, you don't need it anymore.

Mike

On Dec 19, 2008, at 7:36 AM, Cavalera Claudio Luigi wrote:

> Hello guys,
> I'm playing with fs fax capabilities following these guidelines:
> http://wiki.freeswitch.org/wiki/Examples_faxlib.jm
>
> I've compiled mod_fax
> with make mod_fax-install
> and that should have taken care also of spandsp.
>
> When I issue make in scripts/socket2me I get this error:
> socket2me.c:315: error: 'fax_state_t' has no member named 't30_state'
>
> Maybe something has changed in the api?
> Could you please help me track down the problem?
>
> Thanks,
> Claudio
>
>
> Internet Email Confidentiality Footer
> -----------------------------------------------------------------------------------------------------
> La presente comunicazione, con le informazioni in essa contenute e  
> ogni documento o file allegato, e' rivolta unicamente alla/e persona/ 
> e cui e' indirizzata ed alle altre da questa autorizzata/e a  
> riceverla. Se non siete i destinatari/autorizzati siete avvisati che  
> qualsiasi azione, copia, comunicazione, divulgazione o simili basate  
> sul contenuto di tali informazioni e' vietata e potrebbe essere  
> contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia  
> di protezione dei dati personali). Se avete ricevuto questa  
> comunicazione per errore, vi preghiamo di darne immediata notizia al  
> mittente e di distruggere il messaggio originale e ogni file  
> allegato senza farne copia alcuna o riprodurne in alcun modo il  
> contenuto.
>
> This e-mail and its attachments are intended for the addressee(s)  
> only and are confidential and/or may contain legally privileged  
> information. If you have received this message by mistake or are not  
> one of the addressees above, you may take no action based on it, and  
> you may not copy or show it to anyone; please reply to this e-mail  
> and point out the error which has occurred.
> -----------------------------------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From odermann at googlemail.com  Fri Dec 19 08:44:49 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 19 Dec 2008 17:44:49 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
Message-ID: <5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>

it's me again about mod fax... it is short before christmas and my
whish is, to get mod fax working quite reliable. is this possible
under optimal conditions?

all our tests lead by far to more failed faxes than received faxes. i
really like the fax feature and would like to see it beeing usable.

is it just pure luck, if a fax was received or are there some
conditions out there, which could help beeing mod fax reliable?
second question: what about t38? will it come? is there chance, that
it will come? where are the difficulties with mod fax?

our fs servers are standing directly beside the sip switch of our
carrier. from the carriers switch, there is a 50 cm long cat6 cable
going into our cisco-switch. from the cisco switch there are 50 cm
long cat6 cables going into our fs servers.
i doubt, that there can be a signifant packet loss.
are there some settings, we could try out or is the faxing stuff just
unusable, till t38 is supported?



From anthony.minessale at gmail.com  Fri Dec 19 09:00:21 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Dec 2008 11:00:21 -0600
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
	<5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
Message-ID: <191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>

You don't know where the audio goes after that switch in the same room until
it gets to the guy
with the fax machine.

No it will not be improved by Christmas.  Not a chance.

Yes it will probably be much more reliable once it can do T38.

Be happy with what you have for the holiday season.



On Fri, Dec 19, 2008 at 10:44 AM, Dennis <odermann at googlemail.com> wrote:

> it's me again about mod fax... it is short before christmas and my
> whish is, to get mod fax working quite reliable. is this possible
> under optimal conditions?
>
> all our tests lead by far to more failed faxes than received faxes. i
> really like the fax feature and would like to see it beeing usable.
>
> is it just pure luck, if a fax was received or are there some
> conditions out there, which could help beeing mod fax reliable?
> second question: what about t38? will it come? is there chance, that
> it will come? where are the difficulties with mod fax?
>
> our fs servers are standing directly beside the sip switch of our
> carrier. from the carriers switch, there is a 50 cm long cat6 cable
> going into our cisco-switch. from the cisco switch there are 50 cm
> long cat6 cables going into our fs servers.
> i doubt, that there can be a signifant packet loss.
> are there some settings, we could try out or is the faxing stuff just
> unusable, till t38 is supported?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Fri Dec 19 09:09:32 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 19 Dec 2008 18:09:32 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
	<5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
	<191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
Message-ID: <5e414ed0812190909s458f791w65c77c273f2afb80@mail.gmail.com>

hi anthony,

thanks a lot for the clear answer. that is something i can work with :-)

i also want to thank you for the great support you gave us within the
last months and the great freeswitch. our fs servers are up and
running and everything works great (only fax is not working).

have a nice christmas (till i contact you because of some consulting
for final checks ;-)

dennis

2008/12/19 Anthony Minessale <anthony.minessale at gmail.com>:
> You don't know where the audio goes after that switch in the same room until
> it gets to the guy
> with the fax machine.
>
> No it will not be improved by Christmas.  Not a chance.
>
> Yes it will probably be much more reliable once it can do T38.
>
> Be happy with what you have for the holiday season.
>
>
>
> On Fri, Dec 19, 2008 at 10:44 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> it's me again about mod fax... it is short before christmas and my
>> whish is, to get mod fax working quite reliable. is this possible
>> under optimal conditions?
>>
>> all our tests lead by far to more failed faxes than received faxes. i
>> really like the fax feature and would like to see it beeing usable.
>>
>> is it just pure luck, if a fax was received or are there some
>> conditions out there, which could help beeing mod fax reliable?
>> second question: what about t38? will it come? is there chance, that
>> it will come? where are the difficulties with mod fax?
>>
>> our fs servers are standing directly beside the sip switch of our
>> carrier. from the carriers switch, there is a 50 cm long cat6 cable
>> going into our cisco-switch. from the cisco switch there are 50 cm
>> long cat6 cables going into our fs servers.
>> i doubt, that there can be a signifant packet loss.
>> are there some settings, we could try out or is the faxing stuff just
>> unusable, till t38 is supported?
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From odermann at googlemail.com  Fri Dec 19 09:33:02 2008
From: odermann at googlemail.com (Dennis)
Date: Fri, 19 Dec 2008 18:33:02 +0100
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
	<5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
	<191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
Message-ID: <5e414ed0812190933j772292bdw32bbb7213c6b6591@mail.gmail.com>

ahh, just a second. it seems that i did not realize a small
missunderstanding in you answer.

i do not want to SEND a fax, i just want to RECEIVE a fax. so the fax
comes in at out carrier and the rest is sent over about 1m of cat6 to
our fs server.

is there a difference or does it not matter, if we want to receive or
send a fax?





2008/12/19 Anthony Minessale <anthony.minessale at gmail.com>:
> You don't know where the audio goes after that switch in the same room until
> it gets to the guy
> with the fax machine.
>
> No it will not be improved by Christmas.  Not a chance.
>
> Yes it will probably be much more reliable once it can do T38.
>
> Be happy with what you have for the holiday season.
>
>
>
> On Fri, Dec 19, 2008 at 10:44 AM, Dennis <odermann at googlemail.com> wrote:
>>
>> it's me again about mod fax... it is short before christmas and my
>> whish is, to get mod fax working quite reliable. is this possible
>> under optimal conditions?
>>
>> all our tests lead by far to more failed faxes than received faxes. i
>> really like the fax feature and would like to see it beeing usable.
>>
>> is it just pure luck, if a fax was received or are there some
>> conditions out there, which could help beeing mod fax reliable?
>> second question: what about t38? will it come? is there chance, that
>> it will come? where are the difficulties with mod fax?
>>
>> our fs servers are standing directly beside the sip switch of our
>> carrier. from the carriers switch, there is a 50 cm long cat6 cable
>> going into our cisco-switch. from the cisco switch there are 50 cm
>> long cat6 cables going into our fs servers.
>> i doubt, that there can be a signifant packet loss.
>> are there some settings, we could try out or is the faxing stuff just
>> unusable, till t38 is supported?
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mszlazak at aol.com  Fri Dec 19 11:30:42 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Fri, 19 Dec 2008 14:30:42 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
Message-ID: <8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>


?With the firewall ON or OFF the problem still remains.

I've tried 3 different set-ups in a dial plan extension.

1. With? only <action application="set" data="proxy_media=true"/> before bridging.
??? ??? ??? 
2. With only? <action application="set" data="bypass_media=true"/> before bridging.

3. Neither of the above in the extension.

Only 2 with "proxy-media=true" gets the audio across endpoints.

Help :-)


? 




 


 

-----Original Message-----
From: Michael Jerris <mike at jerris.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, 19 Dec 2008 7:49 am
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?









It gives me the impression there is something wrong with your firewall running on the box.



Mike






On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:


I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.
 I'm trying to pass audio "internally" on the same machine between endpoints and have be advised that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 
 However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.
 Any comments?? 
 



Listen to 350+ music, sports, & news radio stations=2
0? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 _______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




=


 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From jason at jasonjgw.net  Fri Dec 19 15:13:00 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Dec 2008 10:13:00 +1100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
Message-ID: <20081219231300.GA4413@jdc.jasonjgw.net>

On Thu, Dec 18, 2008 at 11:20:57AM -0600, Anthony Minessale wrote:
> It seems to be related to 20ms vs 30ms ptime.
> 
> What are the 2 devices and what rev of FS are you on?
> 
> There was more code added in the last few weeks to smooth out this
> occurrence.

This might not be the same issue, but as an additional data point, I can
reliably generate audio drop-outs under revision 10725 as follows - it sounds
somewhat like a jittery network connection, with occasional glitches. Network
problems are not involved, though, as the phone and the machine running
FreeSWITCH are both on my desk, connected via my ADSL router.

On a Snom 320 SIP phone, select G.722 as the first codec.

Call extension 3001 in the default context of the supplied FreeSWITCH
configuration, with nobody else calling into the conference, and listen to the
resulting audio.

Changing the packet size from 20ms to 30ms in the phone's configuration and
repeating the test gave me the same result.

When I recompile FreeSWITCH and re-install, I'll test this again and post a
follow-up if the problem persists.




From brian at freeswitch.org  Fri Dec 19 15:28:44 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Dec 2008 17:28:44 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <20081219231300.GA4413@jdc.jasonjgw.net>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
Message-ID: <11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>

Riddle me this... what firmware are you running?

/b

On Dec 19, 2008, at 5:13 PM, Jason White wrote:

> On a Snom 320 SIP phone, select G.722 as the first codec.
>
> Call extension 3001 in the default context of the supplied FreeSWITCH
> configuration, with nobody else calling into the conference, and  
> listen to the
> resulting audio.




From jason at jasonjgw.net  Fri Dec 19 15:36:59 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Dec 2008 10:36:59 +1100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
Message-ID: <20081219233659.GA5134@jdc.jasonjgw.net>

On Fri, Dec 19, 2008 at 05:28:44PM -0600, Brian West wrote:
> Riddle me this... what firmware are you running?
My apologies - 7.1.30.




From brian at freeswitch.org  Fri Dec 19 15:41:02 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Dec 2008 17:41:02 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <20081219233659.GA5134@jdc.jasonjgw.net>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
Message-ID: <AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>

Please update to 7.1.33 or higher, also I need a pcap of your  
situation email me a link where I can wget it if off list... I need  
the rtp and sip traffic.   You can do it from the phone or from  
FreeSWITCH.
/b

On Dec 19, 2008, at 5:36 PM, Jason White wrote:

> My apologies - 7.1.30.




From brian at freeswitch.org  Fri Dec 19 15:41:28 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Dec 2008 17:41:28 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <20081219233659.GA5134@jdc.jasonjgw.net>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
Message-ID: <2ABAE1B0-B8B1-453D-8B51-661F8E2F5D9B@freeswitch.org>

Btw I have a 300, 320, 360, m3 and an 820 on the way now.  (I don't  
see this problem you're describing at all)

/b

On Dec 19, 2008, at 5:36 PM, Jason White wrote:

> My apologies - 7.1.30.




From jason at jasonjgw.net  Fri Dec 19 15:51:25 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Dec 2008 10:51:25 +1100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <2ABAE1B0-B8B1-453D-8B51-661F8E2F5D9B@freeswitch.org>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
	<2ABAE1B0-B8B1-453D-8B51-661F8E2F5D9B@freeswitch.org>
Message-ID: <20081219235125.GA5269@jdc.jasonjgw.net>

I just tested again, and I myself am now having trouble reproducing it. It
happened on multiple occasions yesterday evening, though.

It is also easier to reproduce with an actual connection to a remote
end-point, but that obviously complicates the situation with potential network
issues. I haven't heard it under G.711 though.

It can't be system load, as the machine has been mostly idle during my
testing/experimentation.




From jason at jasonjgw.net  Fri Dec 19 23:42:35 2008
From: jason at jasonjgw.net (Jason White)
Date: Sat, 20 Dec 2008 18:42:35 +1100
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
	<AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>
Message-ID: <20081220074235.GA4559@jdc.jasonjgw.net>

On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
> Please update to 7.1.33 or higher, also I need a pcap of your  
> situation email me a link where I can wget it if off list... I need  
> the rtp and sip traffic.   You can do it from the phone or from  
> FreeSWITCH.

I've just upgraded FreeSWITCH to 10889, and I'm definitely having audio
issues, but so far only reproduced via remote connections. I'll experiment
with the jitter buffer in case it's network related, and I'll run more local
tests as well.

Yesterday I was experiencing the problem locally, as reported. 

If it persists and if it appears not to be jitter in the network traffic, I'll
run tcpdump and make the output available.

Thanks again for the advice.




From mszlazak at aol.com  Sat Dec 20 00:17:50 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Sat, 20 Dec 2008 03:17:50 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com><B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
Message-ID: <8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>

With the firewall ON or OFF the problem still remains.



I've tried 3 different set-ups in a dial plan extension.



1. With? only <action application="set" data="proxy_media=true"/> before bridging.

??? ??? ??? 

2. With only? <action application="set" data="bypass_media=true"/> before bridging.



3. Neither of the above in the extension.



Only 2 with "bypass-media=true" gets the audio across endpoints.



Help :-)


 


 

-----Original Message-----
From: mszlazak at aol.com
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, 19 Dec 2008 11:30 am
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?











?With the firewall ON or OFF the problem still remains.



I've tried 3 different set-ups in a dial plan extension.



1. With? only <action application="set" data="proxy_media=true"/> before bridging.

??? ??? ??? 

2. With only? <action application="set" data="bypass_media=true"/> before bridging.



3. Neither of the above in the extension.



Only 2 with "proxy-media=true" gets the audio across endpoints.



Help :-)





? 









 





 



-----Original Message-----

From: Michael Jerris <mike at jerris.com>

To: freeswitch-users at lists.freeswitch.org

Sent: Fri, 19 Dec 2008 7:49 am

Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?












It gives me the
 impression there is something wrong with your firewall running on the box.







Mike













On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:



I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.

 I'm trying to pass audio "internally" on the same machine between endpoints and have be advis
ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 

 However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.

 Any comments?? 

 





Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 

 _______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.
org








=



 






_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





 





Listen to 350+ music, sports, & news radio stations ? 
including songs for the holidays ? FREE while you browse. Start Listening Now! 



 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

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From jaugenstine at gmail.com  Sat Dec 20 11:04:54 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Sat, 20 Dec 2008 11:04:54 -0800
Subject: [Freeswitch-users]  mod_java.so load issue
Message-ID: <207e7a5e0812201104l6280ba16g265486f750f10604@mail.gmail.com>

I am installing Freeswitch on Fedora.  I was building/installing the
mod_java.so module and I encountered the following load issue:

2008-12-20 10:34:58 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_java.so
**/usr/local/freeswitch/mod/mod_java.so: invalid ELF header**

Is this a build issue?  I am assuming maybe there is a g++ option that is
set incorrectly but my searches on Google and looking at gcc docs have not
provided a solution.

Jonathan
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From mike at jerris.com  Sat Dec 20 11:33:47 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 20 Dec 2008 14:33:47 -0500
Subject: [Freeswitch-users] mod_java.so load issue
In-Reply-To: <207e7a5e0812201104l6280ba16g265486f750f10604@mail.gmail.com>
References: <207e7a5e0812201104l6280ba16g265486f750f10604@mail.gmail.com>
Message-ID: <F009F62E-F171-449D-9B70-4E1090B1DBF0@jerris.com>

I would suggest cleaning and rebuilding the module.  If that doesn't  
work could we arrange access to the box so I can take a look?

Mike

On Dec 20, 2008, at 2:04 PM, "jonathan augenstine" <jaugenstine at gmail.com 
 > wrote:

> I am installing Freeswitch on Fedora.  I was building/installing the  
> mod_java.so module and I encountered the following load issue:
>
> 2008-12-20 10:34:58 [CRIT] switch_loadable_module.c:839  
> switch_loadable_module_load_file() Error Loading module /usr/local/ 
> freeswitch/mod/mod_java.so
> **/usr/local/freeswitch/mod/mod_java.so: invalid ELF header**
>
> Is this a build issue?  I am assuming maybe there is a g++ option  
> that is set incorrectly but my searches on Google and looking at gcc  
> docs have not provided a solution.
>
> Jonathan
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sat Dec 20 12:34:36 2008
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Dec 2008 14:34:36 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <20081220074235.GA4559@jdc.jasonjgw.net>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
	<AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>
	<20081220074235.GA4559@jdc.jasonjgw.net>
Message-ID: <28CA023D-4A27-48D5-8261-2D7D3A74400E@freeswitch.org>

Do the pcap and show me... because I do not have this issue and I have  
the exact same phone.

/b

On Dec 20, 2008, at 1:42 AM, Jason White wrote:

> On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
>> Please update to 7.1.33 or higher, also I need a pcap of your
>> situation email me a link where I can wget it if off list... I need
>> the rtp and sip traffic.   You can do it from the phone or from
>> FreeSWITCH.
>
> I've just upgraded FreeSWITCH to 10889, and I'm definitely having  
> audio
> issues, but so far only reproduced via remote connections. I'll  
> experiment
> with the jitter buffer in case it's network related, and I'll run  
> more local
> tests as well.
>
> Yesterday I was experiencing the problem locally, as reported.
>
> If it persists and if it appears not to be jitter in the network  
> traffic, I'll
> run tcpdump and make the output available.
>
> Thanks again for the advice.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jason at jasonjgw.net  Sat Dec 20 23:21:11 2008
From: jason at jasonjgw.net (Jason White)
Date: Sun, 21 Dec 2008 18:21:11 +1100
Subject: [Freeswitch-users] error creating IPv6 profile
	when	corresponding IPv4 port in use
In-Reply-To: <20081218091149.GA11826@jdc.jasonjgw.net>
References: <20081217050701.GA14997@jdc.jasonjgw.net>
	<4583E6C2-E037-4DC8-9E4D-CADF203AF6CF@freeswitch.org>
	<20081217051935.GA15357@jdc.jasonjgw.net>
	<20081217074002.GA16365@jdc.jasonjgw.net>
	<5502F5A1-470D-4966-BD0B-3033EB809E84@jerris.com>
	<20081218085323.GA11635@jdc.jasonjgw.net>
	<0B5DCBAA-5FA5-488A-8189-39878A5FCDA0@freeswitch.org>
	<20081218091149.GA11826@jdc.jasonjgw.net>
Message-ID: <20081221072111.GA12815@jdc.jasonjgw.net>

The solution was to edit /etc/asterisk/sip.conf and change
bindaddr = 0.0.0.0
to
bindaddr = x.x.x.x
where x.x.x.x is, naturally, replaced by the host's actual IPv4 address.

Following this change to the Asterisk configuration, Asterisk can bind to port
5060 under IPv4, and FreeSWITCH can bind to port 5060 under IPv6.





From saigop at gmail.com  Sun Dec 21 07:09:41 2008
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Sun, 21 Dec 2008 20:39:41 +0530
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <DB682CA7-D304-4D78-ABA7-4BD91FF6A8AA@jerris.com>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
	<F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
	<2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>
	<DB682CA7-D304-4D78-ABA7-4BD91FF6A8AA@jerris.com>
Message-ID: <2ea4d47e0812210709q59ced653t15bde61cb9b4683e@mail.gmail.com>

Hi Micheal,
     Is it anything like i am violating the laws? please let me know.

On Fri, Dec 5, 2008 at 8:11 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Dec 5, 2008, at 6:23 AM, Gopalakrishnan A.N wrote:
>
> > Hi Micheal,
> >
> >   Thanks for the reply! cant I try with tone detect?
> >
> >   Like dial a number in session and try to detect with tone detect
> > and then bridge the call with some extension.
>
> If you know the exact frequency of the tone you can, but I suspect you
> do not.
>
> Mike
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thank you  with regards,
Gopal,
PeopleTech Systems Private Limited
www.peopletech.co.in
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From saigop at gmail.com  Sun Dec 21 07:12:58 2008
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Sun, 21 Dec 2008 20:42:58 +0530
Subject: [Freeswitch-users] sessions not ending up
Message-ID: <2ea4d47e0812210712o10ac8c28kd1222702d6f34c88@mail.gmail.com>

Hi,
    I have configured the freeswitch, we are dialing through event socket,
if i dial a call per day say around 200 to 300 calls, at the end of the day
the sessions are not ending up in the freeswitch, i can able to see in the
console till all the calls were hanged up, I am using .NET crm.

    Any suggestions would help us.

-- 
Thank you  with regards,
Gopal,
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From msc at freeswitch.org  Sun Dec 21 12:30:59 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Sun, 21 Dec 2008 12:30:59 -0800
Subject: [Freeswitch-users] Predictive Dialing
In-Reply-To: <2ea4d47e0812210709q59ced653t15bde61cb9b4683e@mail.gmail.com>
References: <2ea4d47e0812040450q3ad368dfl55b971a20c9b84ec@mail.gmail.com>
	<F6DAB1E0-9687-4A55-A1D7-362DBB994534@freeswitch.org>
	<2ea4d47e0812050323g1d670d43o94411ef897f31288@mail.gmail.com>
	<DB682CA7-D304-4D78-ABA7-4BD91FF6A8AA@jerris.com>
	<2ea4d47e0812210709q59ced653t15bde61cb9b4683e@mail.gmail.com>
Message-ID: <C71E023A-B1FE-43B6-98D5-A43850117B75@freeswitch.org>

I am not a lawyer so I can't tell you for sure. However, I'm not aware  
of any US laws against beep detection.

-MC

Sent from my iPhone

On Dec 21, 2008, at 7:09 AM, "Gopalakrishnan A.N" <saigop at gmail.com>  
wrote:

> Hi Micheal,
>
>      Is it anything like i am violating the laws? please let me know.
>
> On Fri, Dec 5, 2008 at 8:11 PM, Michael Jerris <mike at jerris.com>  
> wrote:
>
> On Dec 5, 2008, at 6:23 AM, Gopalakrishnan A.N wrote:
>
> > Hi Micheal,
> >
> >   Thanks for the reply! cant I try with tone detect?
> >
> >   Like dial a number in session and try to detect with tone detect
> > and then bridge the call with some extension.
>
> If you know the exact frequency of the tone you can, but I suspect you
> do not.
>
> Mike
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Thank you  with regards,
> Gopal,
> PeopleTech Systems Private Limited
> www.peopletech.co.in
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Sun Dec 21 13:12:51 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 21 Dec 2008 15:12:51 -0600
Subject: [Freeswitch-users] "choppy" voice
In-Reply-To: <28CA023D-4A27-48D5-8261-2D7D3A74400E@freeswitch.org>
References: <d021d3be0812180905i69e9b225kf5089a4ff44cfb7@mail.gmail.com>
	<191c3a030812180920p2f2da629n757becf60d03151f@mail.gmail.com>
	<20081219231300.GA4413@jdc.jasonjgw.net>
	<11A7BC3E-7A54-467E-95A9-AE29A76DB772@freeswitch.org>
	<20081219233659.GA5134@jdc.jasonjgw.net>
	<AC672A35-F5D8-4722-B676-E4A94B817040@freeswitch.org>
	<20081220074235.GA4559@jdc.jasonjgw.net>
	<28CA023D-4A27-48D5-8261-2D7D3A74400E@freeswitch.org>
Message-ID: <191c3a030812211312u24f30d82j874dc918b16c4ffe@mail.gmail.com>

is it only the conference?
what if you call an extension playing a file instead.

On Sat, Dec 20, 2008 at 2:34 PM, Brian West <brian at freeswitch.org> wrote:

> Do the pcap and show me... because I do not have this issue and I have
> the exact same phone.
>
> /b
>
> On Dec 20, 2008, at 1:42 AM, Jason White wrote:
>
> > On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
> >> Please update to 7.1.33 or higher, also I need a pcap of your
> >> situation email me a link where I can wget it if off list... I need
> >> the rtp and sip traffic.   You can do it from the phone or from
> >> FreeSWITCH.
> >
> > I've just upgraded FreeSWITCH to 10889, and I'm definitely having
> > audio
> > issues, but so far only reproduced via remote connections. I'll
> > experiment
> > with the jitter buffer in case it's network related, and I'll run
> > more local
> > tests as well.
> >
> > Yesterday I was experiencing the problem locally, as reported.
> >
> > If it persists and if it appears not to be jitter in the network
> > traffic, I'll
> > run tcpdump and make the output available.
> >
> > Thanks again for the advice.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Sun Dec 21 14:49:14 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 21 Dec 2008 16:49:14 -0600
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
Message-ID: <191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>

Try adding more ip to your box and give each thing it's own dedicated
virtual IP.
Doing everything on the same box can be tricky.


On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:

> With the firewall ON or OFF the problem still remains.
>
> I've tried 3 different set-ups in a dial plan extension.
>
> 1. With  only <action application="set" data="proxy_media=true"/> before
> bridging.
>
> 2. With only  <action application="set" data="bypass_media=true"/> before
> bridging.
>
> 3. Neither of the above in the extension.
>
> Only 2 with "bypass-media=true" gets the audio across endpoints.
>
> Help :-)
>
>
>  -----Original Message-----
> From: mszlazak at aol.com
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, 19 Dec 2008 11:30 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media
> work?
>
>   With the firewall ON or OFF the problem still remains.
>
> I've tried 3 different set-ups in a dial plan extension.
>
> 1. With  only <action application="set" data="proxy_media=true"/> before
> bridging.
>
> 2. With only  <action application="set" data="bypass_media=true"/> before
> bridging.
>
> 3. Neither of the above in the extension.
>
> Only 2 with "proxy-media=true" gets the audio across endpoints.
>
> Help :-)
>
>
>
>
>
> 0A
>
>
>  -----Original Message-----
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, 19 Dec 2008 7:49 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media
> work?
>
>  It gives me the impression there is something wrong with your firewall
> running on the box.
>  Mike
>
>   On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>
> I find it strange that I can have to endpoints get audio went using bypass
> media mode but the audio fails to go across endpoints if I use proxy media
> mode.
> I'm trying to pass audio "internally" on the same machine between endpoints
> and have be advis ed that a reason the audio may fail to be passed is
> because there is some RTP timing and IP address/port issues.
> However, FS has no problem "connecting" ports if i change the mode to
> bypass media. This gives me the impression that something is wrong with FS
> proxy media mode.
> Any comments?
>
> ------------------------------
> Listen to 350+ music, sports, & news radio stations ? including songs for
> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
>  _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch. org <http://www.freeswitch.org>
>
>
>  =
>
> _______________________________________________
>
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> ------------------------------
> Listen to 350+ music, sports, & news radio stations ? including songs for
> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
>
> _______________________________________________
>
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> Listen to 350+ music, sports, & news radio stations ? including songs for
> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From can_man at gmx.de  Sun Dec 21 15:55:54 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 22 Dec 2008 00:55:54 +0100
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one does not
	register
Message-ID: <20081221235554.311330@gmx.net>

Hello,

I have two sipgate.de accounts and would like to attache them both at the same time to freeswitch. 
My problem is that I am not sure how to dial out using the second account. For the first one I can do: 

<action application="bridge" data="sofia/gateway/sipgate.de/$1 at sipgate.de"/>

and everything works, but for the second one I always get: [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g. "sofia/gateway/sipgate2/$1 at sipgate.de" or "sofia/gateway/sipgate.de/$1 at sipgate2"
 
My sip profile looks like this:

<include>
  <gateway name="sipgate.de">
    <param name="realm" value="sipgate.de"/>
    <param name="proxy" value="sipgate.de"/>
    <param name="username" value="******"/>
    <param name="password" value="******"/>
    <param name="extension" value="10000"/>
  </gateway>
</include>

<include>
  <gateway name="sipgate2">
    <param name="realm" value="sipgate.de"/>
    <param name="proxy" value="sipgate.de"/>
    <param name="username" value="******"/>
    <param name="password" value="******"/>
    <param name="extension" value="10000"/>
  </gateway>
</include>

I tried to change gateway name with realm, but no luck.


Thank you very much for your help.

Phil
-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From brian at freeswitch.org  Sun Dec 21 16:56:50 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Dec 2008 18:56:50 -0600
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one does not
	register
In-Reply-To: <20081221235554.311330@gmx.net>
References: <20081221235554.311330@gmx.net>
Message-ID: <3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>

Show me the full extension.

/b

On Dec 21, 2008, at 5:55 PM, can_man at gmx.de wrote:

> <action application="bridge" data="sofia/gateway/sipgate.de/$1 at sipgate.de 
> "/>
>
> and everything works, but for the second one I always get:  
> [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g. "sofia/gateway/sipgate2/$1 at sipgate.de 
> " or "sofia/gateway/sipgate.de/$1 at sipgate2"

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From daldworth at teliax.com  Sun Dec 21 17:37:11 2008
From: daldworth at teliax.com (David Aldworth)
Date: Sun, 21 Dec 2008 18:37:11 -0700
Subject: [Freeswitch-users] Setting codec/dtmf mode
Message-ID: <4F2EBDE9-9EB1-4696-9DFC-651F76BB0EC5@teliax.com>

I'm looking for the most effective way to make sure I'm always forcing  
inband dtmf and PCMU on the PSTN <-> FS side of inbound and outbound  
calls. FS is always in the middle of the media. The FS <-> SIP UA  
(customer) side will be rfc2833 and whatever the negotiated codec for  
that particular UA happens to be. I know I can set <param name="codec- 
prefs" value="PCMU"/> and <param name="inbound-codec-negotiation"  
value="greedy"/> in the internal sip profile but won't the external  
sip profile settings override this when UA dial out? (they hit the  
external profile first in this case)

I'm basically fishing for suggestions on the best way to use start/ 
stop_dtmf for the inband detection and start/stop_dtmf_generate for  
sending the dtmf.

In asterisk this would have been accomplished by setting up separate  
stanza's in sip.conf and setting the dtmfmode= and allow= line per the  
respective legs of the calls. So, calls coming to/from the PSTN would  
have dtmfmode=inband and allow=ulaw, meanwhile UA's connecting to  
asterisk would have dtmfmode=rfc2833 and allow=ulaw, gsm, etc.

Why on earth would I be doing this? Well, in the interest of keeping  
the explanation short, we are limited to the common denominator of all  
our upstream PSTN carriers and they (or their equipment rather) always  
support this setup.

Thanks for any advice.

David




From jason at jasonjgw.net  Sun Dec 21 21:58:16 2008
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Dec 2008 16:58:16 +1100
Subject: [Freeswitch-users] FreeSWITCH port audio module
Message-ID: <20081222055816.GA14532@jdc.jasonjgw.net>

One of the valuable features of FreeSWITCH is that it can be used as a soft
phone, as described on the wiki. In testing this, I discovered the following
issues, any comments on which would be welcome.

1. FreeSWITCH>pa rescan
-ERR no reply

FreeSWITCH>

After plugging in a USB head set and running this command, the new device
wasn't enumerated by pa devlist. Restarting FreeSWITCH of course solved it,
however.

2. More seriously, when using port audio (with a head set as the audio device,
in case that's significant), I'm hearing digital distortion (clipped samples?)
when the other party speaks slightly more loudly than usual. 

When calling via a Snom phone (rather than PortAudio) I have only experienced
this distortion when using FreeSWITCH under G.722, but I'll have to do more
testing to identify the exact combinations that produce it and those which
don't. I haven't heard it with an 8khz call from a SIP phone, so it does
appear to be a FreeSWITCH issue to some extent. I haven't been able to
eliminate it by adjusting the Alsa settings of my audio device.

3. I've encountered errors while trying to access an Intel HDA sound card with
FreeSWITCH, whereby PortAudio fails to open the audio device. Setting the
sample rate in portaudio.conf.xml to 48 khz may have contributed to the
solution, but there have been other changes to my system as well (including a
FreeSWITCh upgrade to revision 10889.

Some sound cards only suport 48 khz, apparently, so if others have problems, I
would suggest adjusting the sample rate in the configuration as I did.




From mszlazak at aol.com  Sun Dec 21 22:44:57 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 22 Dec 2008 01:44:57 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com><B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com><8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com><8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
Message-ID: <8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>


 Hi Anthony,

I actually suggested adding IP's to a Voxeo-Prophecy support person before but they thought that could be problematic. I went along with the earlier warning but now you have suggested it again. What makes everything on the same box tricky? 

Also, the thing that surprises me a bit is that bypass-media works but proxy-media or the default doesn't. 

Would you be kind enough to elaborate.

Thanks. Mark.



 


 

-----Original Message-----
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Sun, 21 Dec 2008 2:49 pm
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?









Try adding more ip to your box and give each thing it's own dedicated virtual IP.
Doing everything on the same box can be tricky.



On Sat, Dec 20, 2008 at 2:17 AM,  <mszlazak at aol.com> wrote:



With the firewall ON or OFF the problem still remains.






I've tried 3 different set-ups in a dial plan extension.





1. With? only <action application="set" data="proxy_media=true"/> before bridging.


??? ??? ??? 


2. With only? <action application="set" data="bypass_media=true"/> before bridging.





3. Neither of the above in the extension.






Only 2 with "bypass-media=true" gets the audio across endpoints.





Help :-)





 





 



-----Original Message-----

From: mszlazak at aol.com

To: freeswitch-users at lists.freeswitch.org




Sent: Fri, 19 Dec 2008 11:30 am

Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?

















?With the firewall ON or OFF the problem still remains.





I've tried 3 different set-ups in a dial plan extension.





1. With? only <action application="set" data="proxy_media=true"/> before bridging.


??? ??? ??? 


2. With only? <action application="set" data="bypass_media=true"/> before bridging.





3. Neither of the above in the extension.





Only 2 with "proxy-media=true" gets the audio across endpoints.





Help :-)








? 










0A







 








 





-----Original Message-----


From: Michael Jerris <mike at jerris.com>


To: freeswitch-users at lists.freeswitch.org


Sent: Fri, 19 Dec 2008 7:49 am


Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?















It gives me the impression there is something wrong with your firewall running on the box.











Mike




















On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:




I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.



 I'm trying to pass audio "internally" on the same machine between endpoints and have be a
dvis
ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 


 However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.


 Any comments?? 


 







Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 



 _______________________________________________



Freeswitch-users mailing list


Freeswitch-users at lists.freeswitch.org


http://lists.freeswitch.org/mailman/listinfo/freeswitch-users


UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users


http://www.freeswitch.
org












=




 







_______________________________________________



Freeswitch-users mailing list



Freeswitch-users at lists.freeswitch.org



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



http://www.freeswitch.org








 








Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 












 






_______________________________________________


Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org






 






Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 





_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org







-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/


AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888

googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From carole.olivier at enst.fr  Sun Dec 21 23:25:33 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Sun, 21 Dec 2008 23:25:33 -0800 (PST)
Subject: [Freeswitch-users] dynamic conference
In-Reply-To: <494A9EE2.7050507@freeswitch.org>
References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com>
	<21054080.post@talk.nabble.com> <49491847.3030403@lists.rupa.com>
	<21069519.post@talk.nabble.com> <494A9EE2.7050507@freeswitch.org>
Message-ID: <21123756.post@talk.nabble.com>


Hello,

Ray, thanks for your detailed answer!!

Fidibus83, when are you pressing *1 ? Because the last version of the
program I gave is thought to be executed like the following:
 - A and B are on call
 - A and B decide to change their call into a conference in order to invite
other people. So A or B press first *2. In this way, both legs are
transferred to the extension named transf_both_legs. In this extension, the
name of the conference will be created and added in the database so you can
not skip it.
 - A and B are now in a conference
 - A (or B) puts B (or A) on hold.
 - A (or B) calls C.
 - if C answers and agrees then A (or B) press *1. This transfers the
opposite leg, this means C, to the extension transf_opposite_leg. C will be
here transferred to the conference whose name will also be added in the db
as an entry for C.
 - A (or B) comes back to the conference it had put on hold.
 - A, B and C are now all in the conference.

So I am not sure but maybe you press first *1 instead of *2 isn't?
If this is not the case then you could press F8 to debug and find out what's
wrong. I am sorry but I can not help you more, I am new with freeswitch and
not very experienced.

Thanks to all of you for your help,
I am going to improve this small program,
Best regards,
Carole


Raymond Chandler-2 wrote:
> 
> Carole O. wrote:
>> Hello,
>>
>> Thanks for your answers!
>> Concerning the creation of a new variable for the conference the problem
>> is
>> that I do not create channels from the conference. I call separately a
>> new
>> member on a new channel and add it on the conference only if he agrees to
>> enter it. So it was the same problem as for the uuid, I am not sure I can
>> access the good variable from anywhere in case many conferences are
>> running.
>>
>>   
> you could use the db app to hold state across multiple calls... maybe 
> use the ${caller_id_number} and the ${destination_number} as keys for 
> the insert/select so that there's something constant to use in the 
> select... and another extension or two may be needed...
> 
> You could do the db lookup before you make the call so that you see if 
> your caller is already a member of a conference.... if he is, then the 
> transfer from *1 would work much the same as it does now except you'd 
> use the result of the db lookup as the conference number... if he's not 
> a member of an existing conference, then you could generate the uuid 
> like Anthony said before, then do a db insert for ${caller_id_number} 
> and ${destination_number} to insert that newly created uuid and use it 
> as the conference number.... one caveat that i see here is that the 
> destination_number would have to be exactly the same as if that user 
> were callling and it was his caller_id_number, otherwise your query will 
> fail.
> 
> you'll also need to "clean" the db when you hangup, which should be able 
> to be accomplished with an execute_on_hangup that does a delete of the 
> conf data for each user
> 
> -Ray
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/dynamic-conference-tp21053181p21123756.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From Claudio.Cavalera at italtel.it  Mon Dec 22 00:28:01 2008
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Mon, 22 Dec 2008 09:28:01 +0100
Subject: [Freeswitch-users] Problem compiling socket2me
In-Reply-To: <94326D53-5AFD-4B71-845D-41F486D02D10@jerris.com>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC0969643@BESONE.corp.dom>

freeswitch-users-bounces at lists.freeswitch.org wrote:
> mod_fax replaces socket2me, you don't need it anymore.
> 
> Mike

Ok thanks,
I would suggest to remove socket2me from trunk if still present.
Ciao,
Claudio


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From can_man at gmx.de  Mon Dec 22 03:06:02 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 22 Dec 2008 12:06:02 +0100
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one does
	not	register
In-Reply-To: <3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
Message-ID: <20081222110602.63410@gmx.net>



> Show me the full extension.

This extension with sipgate.de works for the single number specified, however when I change it to sipgate2 it doesn't. 

   <extension name="Long Distance">
    <condition field="destination_number" expression="^(0708351963)$">
     <action application="set" data="effective_caller_id_name=John Freeswitch"/>
     <action application="set" data="effective_caller_id_number=07083970139"/>
     <action application="bridge" data="sofia/gateway/sipgate.de/$1 at sipgate.de"/>
    </condition>
   </extension>

Thanks for your help.

Phil

> 
> On Dec 21, 2008, at 5:55 PM, can_man at gmx.de wrote:
> 
> > <action application="bridge"
> data="sofia/gateway/sipgate.de/$1 at sipgate.de 
> > "/>
> >
> > and everything works, but for the second one I always get:  
> > [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g.
> "sofia/gateway/sipgate2/$1 at sipgate.de 
> > " or "sofia/gateway/sipgate.de/$1 at sipgate2"
> 

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From pieter_eduard at biznetnetworks.com  Mon Dec 22 05:19:38 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Mon, 22 Dec 2008 20:19:38 +0700
Subject: [Freeswitch-users] call failed from PLMN to enum number
Message-ID: <494F93EA.4080608@biznetnetworks.com>

Hi,

I have an enum number, if I call the number from any ip extension ( i 
use default enum.conf that points to e164.arpa)  then the call goes well 
to my ATA that registers to my fs box,
but if i try to call the number from PLMN, i get the ring at my ATA and 
if i pick it up, there's no sound.

here's my public.xml config :

 <extension name="enum">
        <condition field="destination_number" expression="^12345678">
        <action application="set" data="proxy_media=true"/>
        <action application="bridge" data="sofia/default/1000%1.2.3.4"/>
        </condition>
    </extension>

For more detailed debug log, i already submit it on jira : 
http://jira.freeswitch.org/browse/MODAPP-186


regards,

-Pieter-




From carole.olivier at enst.fr  Mon Dec 22 05:35:44 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Mon, 22 Dec 2008 05:35:44 -0800 (PST)
Subject: [Freeswitch-users]  close channels properly
Message-ID: <21127913.post@talk.nabble.com>


Hello,

I use the following code to call VoIP speakers and make an announcement:
<extension name="security_announcement">
       <condition field="destination_number" expression="^9001$">
       <action application="set"
data="conference_auto_outcall_caller_id_name=security_announcement"/>
       <action application="set"
data="conference_auto_outcall_id_number=9001"/>
       <action application="set" data="conference_auto_outcall_timeout=60"/>
       <action application="set" data="conference_auto_outcall_flags=mute"/>
       <action application="set"
data="conference_auto_outcall_announce=announcement/announcement_security_delay.wav"/>
       <action application="conference_set_auto_outcall"
data="user/1021@$${domain}"/>
       <action application="conference_set_auto_outcall"
data="user/1022@$${domain}"/>
       <action application="conference"
data="conf_speaker at profilespeaker_announcement"/>
       </condition>
     </extension>

1021 and 1022 are the speakers.
At the end of the announcement, since there is no noise anymore, the
speakers stop listening but they do not send any messages to tell Freeswitch
it can close the opened channels. The channels are closed only after a
timeout of 5 minutes.

Does anybody know how I could force freeswitch to close all the channels
after the announcement? I have seen there is the application sched_hangup
but when I used it it only closes the channel to the caller and not the
other ones.

Thanks a lot for your help and Happy Holidays!
Carole
-- 
View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21127913.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From can_man at gmx.de  Mon Dec 22 06:05:32 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 22 Dec 2008 15:05:32 +0100
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one
	does	not	register RESOLVED
In-Reply-To: <20081222110602.63410@gmx.net>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
	<20081222110602.63410@gmx.net>
Message-ID: <20081222140532.264490@gmx.net>

Hello,

I know, no top posting, but I want to say that I resolved my problem.
Don't ask me why exactly it works now, but after I removed the effective caller_id_number it works like  this:


   <extension name="Long Distance">
    <condition field="destination_number" expression="^(0708351963)$">
    <action application="bridge" data="sofia/gateway/sipgate2/$1 at sipgate.de"/>
    </condition>
   </extension>


<include>
  <gateway name="sipgate.de">
    <param name="realm" value="sipgate.de"/>
    <param name="proxy" value="sipgate.de"/>
    <param name="username" value="******"/>
    <param name="password" value="******"/>
    <param name="extension" value="10000"/>
    <param name="register" value="true"/>
  </gateway>
</include>

<include>
  <gateway name="sipgate2">
    <param name="realm" value="sipgate.de"/>
    <param name="proxy" value="sipgate.de"/>
    <param name="username" value="******"/>
    <param name="password" value="******"/>
    <param name="extension" value="10000"/>
    <param name="register" value="true"/>
  </gateway>
</include>


I have added a note to the wiki Tested_Phone_Providers_Listing under Sipgate.de


Phil


> 
> 
> > Show me the full extension.
> 
> This extension with sipgate.de works for the single number specified,
> however when I change it to sipgate2 it doesn't. 
> 
>    <extension name="Long Distance">
>     <condition field="destination_number" expression="^(0708351963)$">
>      <action application="set" data="effective_caller_id_name=John
> Freeswitch"/>
>      <action application="set"
> data="effective_caller_id_number=07083970139"/>
>      <action application="bridge"
> data="sofia/gateway/sipgate.de/$1 at sipgate.de"/>
>     </condition>
>    </extension>
> 
> Thanks for your help.
> 
> Phil
> 
> > 
> > On Dec 21, 2008, at 5:55 PM, can_man at gmx.de wrote:
> > 
> > > <action application="bridge"
> > data="sofia/gateway/sipgate.de/$1 at sipgate.de 
> > > "/>
> > >
> > > and everything works, but for the second one I always get:  
> > > [INVALID_NUMBER_FORMAT] Whatever combination I try. E.g.
> > "sofia/gateway/sipgate2/$1 at sipgate.de 
> > > " or "sofia/gateway/sipgate.de/$1 at sipgate2"
> > 
> 
> -- 
> Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen:
> http://www.gmx.net/de/go/multimessenger
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From brian at freeswitch.org  Mon Dec 22 06:35:23 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:35:23 -0600
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <494F93EA.4080608@biznetnetworks.com>
References: <494F93EA.4080608@biznetnetworks.com>
Message-ID: <6FFEA244-18DB-4D39-AC63-AE2AEA1FAFCD@freeswitch.org>

First off I would try the latest SVN Trunk you're a bit behind.  
Secondly I would try without proxy media mode on.

/b

On Dec 22, 2008, at 7:19 AM, Pieter Eduard wrote:

> Hi,
>
> I have an enum number, if I call the number from any ip extension ( i
> use default enum.conf that points to e164.arpa)  then the call goes  
> well
> to my ATA that registers to my fs box,
> but if i try to call the number from PLMN, i get the ring at my ATA  
> and
> if i pick it up, there's no sound.
>
> here's my public.xml config :
>
> <extension name="enum">
>        <condition field="destination_number" expression="^12345678">
>        <action application="set" data="proxy_media=true"/>
>        <action application="bridge" data="sofia/default/ 
> 1000%1.2.3.4"/>
>        </condition>
>    </extension>
>
> For more detailed debug log, i already submit it on jira :
> http://jira.freeswitch.org/browse/MODAPP-186
>
>
> regards,
>
> -Pieter-
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Mon Dec 22 06:36:22 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:36:22 -0600
Subject: [Freeswitch-users] Problem compiling socket2me
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0969643@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0969643@BESONE.corp.dom>
Message-ID: <1104ACEC-491C-4CE6-B8CF-F7A01EC94618@freeswitch.org>

Its there to serve as an example of using the socket interface with  
audio if anything it should be updated to the latest SpanDSP code.  
Patches welcome.

/b

On Dec 22, 2008, at 2:28 AM, Cavalera Claudio Luigi wrote:

> Ok thanks,
> I would suggest to remove socket2me from trunk if still present.
> Ciao,
> Claudio




From brian at freeswitch.org  Mon Dec 22 06:37:17 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:37:17 -0600
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one does
	not	register
In-Reply-To: <20081222110602.63410@gmx.net>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
	<20081222110602.63410@gmx.net>
Message-ID: <CEDD95AF-BA95-48F1-8C6E-54B4536C68DF@freeswitch.org>

When dialing via a gateway DO NOT add the @sipgate.de to it.  That  
will cause it to fail.

/b

On Dec 22, 2008, at 5:06 AM, can_man at gmx.de wrote:

>
>   <extension name="Long Distance">
>    <condition field="destination_number" expression="^(0708351963)$">
>     <action application="set" data="effective_caller_id_name=John  
> Freeswitch"/>
>     <action application="set"  
> data="effective_caller_id_number=07083970139"/>
>     <action application="bridge" data="sofia/gateway/sipgate.de/$1 at sipgate.de 
> "/>
>    </condition>
>   </extension>

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From brian at freeswitch.org  Mon Dec 22 06:38:36 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:38:36 -0600
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one
	does	not	register RESOLVED
In-Reply-To: <20081222140532.264490@gmx.net>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
	<20081222110602.63410@gmx.net> <20081222140532.264490@gmx.net>
Message-ID: <66CA1BC2-9099-4B3D-82C3-18F3B013B521@freeswitch.org>

Remove the @sipgate.de it won't work properly if you add that.. you  
have this setting in the gateway xml so don't add it to the bridge  
line too.

/b

On Dec 22, 2008, at 8:05 AM, can_man at gmx.de wrote:

>    <action application="bridge" data="sofia/gateway/sipgate2/$1 at sipgate.de 
> "/>

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From brian at freeswitch.org  Mon Dec 22 06:39:17 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:39:17 -0600
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <21127913.post@talk.nabble.com>
References: <21127913.post@talk.nabble.com>
Message-ID: <2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>

What do you mean they close (hangup) after the 5 minute timeout?

/b

On Dec 22, 2008, at 7:35 AM, Carole O. wrote:

> 1021 and 1022 are the speakers.
> At the end of the announcement, since there is no noise anymore, the
> speakers stop listening but they do not send any messages to tell  
> Freeswitch
> it can close the opened channels. The channels are closed only after a
> timeout of 5 minutes.
>
> Does anybody know how I could force freeswitch to close all the  
> channels
> after the announcement? I have seen there is the application  
> sched_hangup
> but when I used it it only closes the channel to the caller and not  
> the
> other ones.




From kristjan.ugrin at gmail.com  Mon Dec 22 06:42:08 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Mon, 22 Dec 2008 15:42:08 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
Message-ID: <op.umki4gzjatcbto@linux.site>

I modified mod_dingaling.c so I can intercept google talk chat messages  
via socket - nothing fancy.
Then I wrote a java app that connects to freeswitch socket and in case of  
a proper message (trigger) it sends a command to freeswitch, in my case:
api originate sofia/default/1001 at 10.99.8.221  
&bridge(dingaling/gmail.com/my_mail at gmail.com)

Dingaling is logged in as another user which I have added as buddy, chat  
messages go trough, however when a call is started
between SIP and Gtalk client, we cannot hear each other at all.
Using freeswitch revision: 10866

Here is the log:
http://pastebin.com/m1eba2cb8

What can be the problem? First I thought it is because running sip client  
+ gtalk and freeswitch on one host, but then I
moved SIP phone and Gtalk to 2 different workstations, using the third  
only for freeswitch. Also calls from "call" example program
 from google lib works fine with same setup - something must be problematic  
with freeswitch, however cannot see what.

Thank you!

-- 
kriko



From brian at freeswitch.org  Mon Dec 22 06:44:51 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 08:44:51 -0600
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <494F93EA.4080608@biznetnetworks.com>
References: <494F93EA.4080608@biznetnetworks.com>
Message-ID: <CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>

I also need you to do this call again with "console loglevel debug" on  
and post it attached to the jira and not inline on the comments please.

/b
On Dec 22, 2008, at 7:19 AM, Pieter Eduard wrote:

> Hi,
>
> I have an enum number, if I call the number from any ip extension ( i
> use default enum.conf that points to e164.arpa)  then the call goes  
> well
> to my ATA that registers to my fs box,
> but if i try to call the number from PLMN, i get the ring at my ATA  
> and
> if i pick it up, there's no sound.
>
> here's my public.xml config :
>
> <extension name="enum">
>        <condition field="destination_number" expression="^12345678">
>        <action application="set" data="proxy_media=true"/>
>        <action application="bridge" data="sofia/default/ 
> 1000%1.2.3.4"/>
>        </condition>
>    </extension>
>
> For more detailed debug log, i already submit it on jira :
> http://jira.freeswitch.org/browse/MODAPP-186
>
>
> regards,
>
> -Pieter-
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From can_man at gmx.de  Mon Dec 22 06:52:56 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 22 Dec 2008 15:52:56 +0100
Subject: [Freeswitch-users] 2 Sipgate.de accounts - second one
	does	not	register
In-Reply-To: <CEDD95AF-BA95-48F1-8C6E-54B4536C68DF@freeswitch.org>
References: <20081221235554.311330@gmx.net>
	<3BDBB22A-3A7B-40E7-9048-19725DA9F421@freeswitch.org>
	<20081222110602.63410@gmx.net>
	<CEDD95AF-BA95-48F1-8C6E-54B4536C68DF@freeswitch.org>
Message-ID: <20081222145256.184760@gmx.net>



> When dialing via a gateway DO NOT add the @sipgate.de to it.  That  
> will cause it to fail.
> 

Thank you. I have also updated the wiki.

Phil



-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From anthony.minessale at gmail.com  Mon Dec 22 07:19:08 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 09:19:08 -0600
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.umki4gzjatcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
Message-ID: <191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>

Your log shows rtp streams being allocated.
did you look at at the packets on the wire with a packet capture program?

You are better off using proper jingle and component mode.  What you are
describing sounds like
a workaround to avoid doing it right.



On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> I modified mod_dingaling.c so I can intercept google talk chat messages
> via socket - nothing fancy.
> Then I wrote a java app that connects to freeswitch socket and in case of
> a proper message (trigger) it sends a command to freeswitch, in my case:
> api originate sofia/default/1001 at 10.99.8.221
> &bridge(dingaling/gmail.com/my_mail at gmail.com)
>
> Dingaling is logged in as another user which I have added as buddy, chat
> messages go trough, however when a call is started
> between SIP and Gtalk client, we cannot hear each other at all.
> Using freeswitch revision: 10866
>
> Here is the log:
> http://pastebin.com/m1eba2cb8
>
> What can be the problem? First I thought it is because running sip client
> + gtalk and freeswitch on one host, but then I
> moved SIP phone and Gtalk to 2 different workstations, using the third
> only for freeswitch. Also calls from "call" example program
>  from google lib works fine with same setup - something must be problematic
> with freeswitch, however cannot see what.
>
> Thank you!
>
> --
> kriko
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec 22 07:24:53 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 09:24:53 -0600
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
	<8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
Message-ID: <191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>

I don't really know what your problem is.  I just saw you ask 3 times for
help and tried to offer a suggestion.
if you start FS with TPORT_LOG=1 you can see all the sip messages in the
console and you could
also run wireshark to look at a packet capture.

If you use the same IP for media on the same box for 3 programs at once you
may end up with 2 applictions choosing the same media port etc.

It's just a good practice to run every voip program on it's own IP.




On Mon, Dec 22, 2008 at 12:44 AM, <mszlazak at aol.com> wrote:

>  Hi Anthony,
>
> I actually suggested adding IP's to a Voxeo-Prophecy support person before
> but they thought that could be problematic. I went along with the earlier
> warning but now you have suggested it again. What makes everything on the
> same box tricky?
>
> Also, the thing that surprises me a bit is that bypass-media works but
> proxy-media or the default doesn't.
>
> Would you be kind enough to elaborate.
>
> Thanks. Mark.
>
>
>
>  -----Original Message-----
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sun, 21 Dec 2008 2:49 pm
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media
> work?
>
>  Try adding more ip to your box and give each thing it's own dedicated
> virtual IP.
> Doing everything on the same box can be tricky.
>
>
> On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:
>
>> With the firewall ON or OFF the problem still remains.
>>
>> I've tried 3 different set-ups in a dial plan extension.
>>
>> 1. With  only <action application="set" data="proxy_media=true"/> before
>> bridging.
>>
>> 2. With only  <action application="set" data="bypass_media=true"/> before
>> bridging.
>>
>> 3. Neither of the above in the extension.
>>
>>  Only 2 with "bypass-media=true" gets the audio across endpoints.
>>
>> Help :-)
>>
>>
>>  -----Original Message-----
>> From: mszlazak at aol.com
>> To: freeswitch-users at lists.freeswitch.org
>>  Sent: Fri, 19 Dec 2008 11:30 am
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>> Media work?
>>
>>    With the firewall ON or OFF the problem still remains.
>>
>> I've tried 3 different set-ups in a dial plan extension.
>>
>> 1. With  only <action application="set" data="proxy_media=true"/> before
>> bridging.
>>
>> 2. With only  <action application="set" data="bypass_media=true"/> before
>> bridging.
>>
>> 3. Neither of the above in the extension.
>>
>> Only 2 with "proxy-media=true" gets the audio across endpoints.
>>
>> Help :-)
>>
>>
>>
>>
>>
>>  0A
>>
>>
>>  -----Original Message-----
>> From: Michael Jerris <mike at jerris.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Fri, 19 Dec 2008 7:49 am
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>> Media work?
>>
>>  It gives me the impression there is something wrong with your firewall
>> running on the box.
>>  Mike
>>
>>   On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>>
>> I find it strange that I can have to endpoints get audio went using bypass
>> media mode but the audio fails to go across endpoints if I use proxy media
>> mode.
>> I'm trying to pass audio "internally" on the same machine between
>> endpoints and have be advis ed that a reason the audio may fail to be passed
>> is because there is some RTP timing and IP address/port issues.
>> However, FS has no problem "connecting" ports if i change the mode to
>> bypass media. This gives me the impression that something is wrong with FS
>> proxy media mode.
>> Any comments?
>>
>> ------------------------------
>> Listen to 350+ music, sports, & news radio stations ? including songs for
>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>
>>  _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch. org <http://www.freeswitch.org>
>>
>>
>>  =
>>
>> _______________________________________________
>>
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> ------------------------------
>> Listen to 350+ music, sports, & news radio stations ? including songs for
>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>
>>
>> _______________________________________________
>>
>>
>>
>> Freeswitch-users mailing list
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> Listen to 350+ music, sports, & news radio stations ? including songs for
>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<http://lists.freeswitch.org/mailman/listinfo++/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
>
> Freeswitch-users mailing list
> Freeswitch-users at lists
> .freeswitch.org <Freeswitch-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> Listen to 350+ music, sports, & news radio stations ? including songs for
> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From ser at man.szczecin.pl  Mon Dec 22 05:54:40 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Mon, 22 Dec 2008 14:54:40 +0100
Subject: [Freeswitch-users] group call with BLF and pickup
Message-ID: <1229954080.9989.41.camel@worek.man.szczecin.pl>

Hi,

I'm currently evaluating FreeSWITCH to see if I can migrate to it from
Asterisk (which gives me now all functions I need, but has some problems
with predictability). I need only one atypical functionality:
1. we have one main extension, which clients call. lets name it 5555
2. when someone calls 5555, then immediately rings only one
   phone (1001) and 15 seconds later 3 more phones (1002-1004). phone
   1001 should be dialled once per call to 5555
3. all phones (1000-1010) have BLF monitoring 5555, so anyone can 
   pick it up when someone calls it

ad 3. it is acceptable that all phones monitor 1001 instead of 5555 to
pick up connection to 5555

After reading and googling about possible implementation in FS for few
hours I couldn't find anything useful. 

Could you tell if it is possible to implement in FS without artificial
limbs (i dunno if it's best English word for what I mean :) like I had
to do in Asterisk?

greetings,
Seweryn





From anthony.minessale at gmail.com  Mon Dec 22 07:39:57 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 09:39:57 -0600
Subject: [Freeswitch-users] FreeSWITCH port audio module
In-Reply-To: <20081222055816.GA14532@jdc.jasonjgw.net>
References: <20081222055816.GA14532@jdc.jasonjgw.net>
Message-ID: <191c3a030812220739y361b69e1kd45b9258733aff23@mail.gmail.com>

You did not clarify any of the details of your machine HW, OS etc.
The issue you reported with snoms are not reproducible based on your
description so you should
try harded to pinpoint it.  Telling us you have a problem but you are not
sure why etc is not helpful because now I have to stop and wait for you to
reply if I want to help you further.
Portaudio seems to not work very well under linux and we don't have any
linux machines with sound to test it with.

Can you please provide all the details including possibly access to your
machine so we can
reproduce your issue with your machine if we can't with our own.

Consider joining irc and informing us in real time

irc.freenode.net #freeswitch



On Sun, Dec 21, 2008 at 11:58 PM, Jason White <jason at jasonjgw.net> wrote:

> One of the valuable features of FreeSWITCH is that it can be used as a soft
> phone, as described on the wiki. In testing this, I discovered the
> following
> issues, any comments on which would be welcome.
>
> 1. FreeSWITCH>pa rescan
> -ERR no reply
>
> FreeSWITCH>
>
> After plugging in a USB head set and running this command, the new device
> wasn't enumerated by pa devlist. Restarting FreeSWITCH of course solved it,
> however.
>
> 2. More seriously, when using port audio (with a head set as the audio
> device,
> in case that's significant), I'm hearing digital distortion (clipped
> samples?)
> when the other party speaks slightly more loudly than usual.
>
> When calling via a Snom phone (rather than PortAudio) I have only
> experienced
> this distortion when using FreeSWITCH under G.722, but I'll have to do more
> testing to identify the exact combinations that produce it and those which
> don't. I haven't heard it with an 8khz call from a SIP phone, so it does
> appear to be a FreeSWITCH issue to some extent. I haven't been able to
> eliminate it by adjusting the Alsa settings of my audio device.
>
> 3. I've encountered errors while trying to access an Intel HDA sound card
> with
> FreeSWITCH, whereby PortAudio fails to open the audio device. Setting the
> sample rate in portaudio.conf.xml to 48 khz may have contributed to the
> solution, but there have been other changes to my system as well (including
> a
> FreeSWITCh upgrade to revision 10889.
>
> Some sound cards only suport 48 khz, apparently, so if others have
> problems, I
> would suggest adjusting the sample rate in the configuration as I did.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristjan.ugrin at gmail.com  Mon Dec 22 07:42:17 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Mon, 22 Dec 2008 16:42:17 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
Message-ID: <op.umklwrrxatcbto@linux.site>

There are absolutely no UDP packets going trough like when doing a call  
 from gtalk to gtalk.

You mean this (component mode):
http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F
Is there more documentation that this?

All I would like to do is to initiate a call between SIP telephone and  
gtalk user who typed in the message.

Thank you!


On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale  
<anthony.minessale at gmail.com> wrote:

> Your log shows rtp streams being allocated.
> did you look at at the packets on the wire with a packet capture program?
>
> You are better off using proper jingle and component mode.  What you are
> describing sounds like
> a workaround to avoid doing it right.
>
>
>
> On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> I modified mod_dingaling.c so I can intercept google talk chat messages
>> via socket - nothing fancy.
>> Then I wrote a java app that connects to freeswitch socket and in case  
>> of
>> a proper message (trigger) it sends a command to freeswitch, in my case:
>> api originate sofia/default/1001 at 10.99.8.221
>> &bridge(dingaling/gmail.com/my_mail at gmail.com)
>>
>> Dingaling is logged in as another user which I have added as buddy, chat
>> messages go trough, however when a call is started
>> between SIP and Gtalk client, we cannot hear each other at all.
>> Using freeswitch revision: 10866
>>
>> Here is the log:
>> http://pastebin.com/m1eba2cb8
>>
>> What can be the problem? First I thought it is because running sip  
>> client
>> + gtalk and freeswitch on one host, but then I
>> moved SIP phone and Gtalk to 2 different workstations, using the third
>> only for freeswitch. Also calls from "call" example program
>>  from google lib works fine with same setup - something must be  
>> problematic
>> with freeswitch, however cannot see what.
>>
>> Thank you!
>>
>> --
>> kriko
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>



-- 
Porn - the reason you need a new hard drive.



From gilbertandrew at me.com  Mon Dec 22 07:49:14 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Mon, 22 Dec 2008 10:49:14 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
 work?
In-Reply-To: <191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
	<8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
	<191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
Message-ID: <8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>

Mark,

Sorry I haven't had much time to help with this either.

But Anthony is offering good advice here. You are either going to have  
to work out what is going on at SIP/SDP/RTP level through logs and  
wireshark, or opt for a separate ip space. Another option (besides  
virtual ips) is VMWare or VirtualBox, although VMWare is probably  
easier to setup and bridge naturally to your host.

Vm's are just so easy anymore and it definitely seems like you are  
going against the grain right now.

Also - realizing you got here because of the need for ASR. I do have  
the Lumenvox license, and I was able to compile the module out of SVN.  
I have not tested anything yet. If things go well I should have some  
time after the 25th for this. My goal would be to get pizza or  
something akin to work.

Andy


On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:

> I don't really know what your problem is.  I just saw you ask 3  
> times for help and tried to offer a suggestion.
> if you start FS with TPORT_LOG=1 you can see all the sip messages in  
> the console and you could
> also run wireshark to look at a packet capture.
>
> If you use the same IP for media on the same box for 3 programs at  
> once you may end up with 2 applictions choosing the same media port  
> etc.
>
> It's just a good practice to run every voip program on it's own IP.
>
>
>
>
> On Mon, Dec 22, 2008 at 12:44 AM, <mszlazak at aol.com> wrote:
> Hi Anthony,
>
> I actually suggested adding IP's to a Voxeo-Prophecy support person  
> before but they thought that could be problematic. I went along with  
> the earlier warning but now you have suggested it again. What makes  
> everything on the same box tricky?
>
> Also, the thing that surprises me a bit is that bypass-media works  
> but proxy-media or the default doesn't.
>
> Would you be kind enough to elaborate.
>
> Thanks. Mark.
>
>
>
> -----Original Message-----
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sun, 21 Dec 2008 2:49 pm
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
> Proxy Media work?
>
> Try adding more ip to your box and give each thing it's own  
> dedicated virtual IP.
> Doing everything on the same box can be tricky.
>
>
> On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:
> With the firewall ON or OFF the problem still remains.
>
> I've tried 3 different set-ups in a dial plan extension.
>
> 1. With  only <action application="set" data="proxy_media=true"/>  
> before bridging.
>
> 2. With only  <action application="set" data="bypass_media=true"/>  
> before bridging.
>
> 3. Neither of the above in the extension.
>
> Only 2 with "bypass-media=true" gets the audio across endpoints.
>
> Help :-)
>
>
> -----Original Message-----
> From: mszlazak at aol.com
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, 19 Dec 2008 11:30 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
> Proxy Media work?
>
>  With the firewall ON or OFF the problem still remains.
>
> I've tried 3 different set-ups in a dial plan extension.
>
> 1. With  only <action application="set" data="proxy_media=true"/>  
> before bridging.
>
> 2. With only  <action application="set" data="bypass_media=true"/>  
> before bridging.
>
> 3. Neither of the above in the extension.
>
> Only 2 with "proxy-media=true" gets the audio across endpoints.
>
> Help :-)
>
>
>
>
>
> 0A
>
>
> -----Original Message-----
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Fri, 19 Dec 2008 7:49 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
> Proxy Media work?
>
> It gives me the impression there is something wrong with your  
> firewall running on the box.
>
> Mike
>
> On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>
>> I find it strange that I can have to endpoints get audio went using  
>> bypass media mode but the audio fails to go across endpoints if I  
>> use proxy media mode.
>> I'm trying to pass audio "internally" on the same machine between  
>> endpoints and have be advis ed that a reason the audio may fail to  
>> be passed is because there is some RTP timing and IP address/port  
>> issues.
>> However, FS has no problem "connecting" ports if i change the mode  
>> to bypass media. This gives me the impression that something is  
>> wrong with FS proxy media mode.
>> Any comments?
>>
>> Listen to 350+ music, sports, & news radio stations ? including  
>> songs for the holidays ? FREE while you browse. Start Listening  
>> Now!
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch. org
>
> =
> _______________________________________________
>
>
>
>
>
>
>
> Freeswitch-users mailing list
>
>
>
>
>
>
>
> Freeswitch-users at lists.freeswitch.org
>
>
>
>
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
>
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
>
>
>
>
> http://www.freeswitch.org
>
>
>
>
>
>
>
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
>  _______________________________________________
>
>
>
>
>
>
> Freeswitch-users mailing list
>
>
>
> Freeswitch-users at lists.freeswitch.org
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
>
> http://www.freeswitch.org
>
>
>
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
>
> Freeswitch-users mailing list
>
> Freeswitch-users at lists
> .freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Dec 22 07:55:52 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 09:55:52 -0600
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
	<8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
	<191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
	<8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
Message-ID: <191c3a030812220755k6141aa45g219039fa20d3b66f@mail.gmail.com>

Andy,

ping us when you are ready. we have a lumenvox version of the pizza thing
already
That's the one we started with.


On Mon, Dec 22, 2008 at 9:49 AM, Andrew Gilbert <gilbertandrew at me.com>wrote:

> Mark,
> Sorry I haven't had much time to help with this either.
>
> But Anthony is offering good advice here. You are either going to have to
> work out what is going on at SIP/SDP/RTP level through logs and wireshark,
> or opt for a separate ip space. Another option (besides virtual ips) is
> VMWare or VirtualBox, although VMWare is probably easier to setup and bridge
> naturally to your host.
>
> Vm's are just so easy anymore and it definitely seems like you are going
> against the grain right now.
>
> Also - realizing you got here because of the need for ASR. I do have the
> Lumenvox license, and I was able to compile the module out of SVN. I have
> not tested anything yet. If things go well I should have some time after the
> 25th for this. My goal would be to get pizza or something akin to work.
>
> Andy
>
>
> On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:
>
> I don't really know what your problem is.  I just saw you ask 3 times for
> help and tried to offer a suggestion.
> if you start FS with TPORT_LOG=1 you can see all the sip messages in the
> console and you could
> also run wireshark to look at a packet capture.
>
> If you use the same IP for media on the same box for 3 programs at once you
> may end up with 2 applictions choosing the same media port etc.
>
> It's just a good practice to run every voip program on it's own IP.
>
>
>
>
> On Mon, Dec 22, 2008 at 12:44 AM, <mszlazak at aol.com> wrote:
>
>>  Hi Anthony,
>>
>> I actually suggested adding IP's to a Voxeo-Prophecy support person before
>> but they thought that could be problematic. I went along with the earlier
>> warning but now you have suggested it again. What makes everything on the
>> same box tricky?
>>
>> Also, the thing that surprises me a bit is that bypass-media works but
>> proxy-media or the default doesn't.
>>
>> Would you be kind enough to elaborate.
>>
>> Thanks. Mark.
>>
>>
>>
>>  -----Original Message-----
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Sun, 21 Dec 2008 2:49 pm
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>> Media work?
>>
>>  Try adding more ip to your box and give each thing it's own dedicated
>> virtual IP.
>> Doing everything on the same box can be tricky.
>>
>>
>> On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:
>>
>>> With the firewall ON or OFF the problem still remains.
>>>
>>> I've tried 3 different set-ups in a dial plan extension.
>>>
>>> 1. With  only <action application="set" data="proxy_media=true"/> before
>>> bridging.
>>>
>>> 2. With only  <action application="set" data="bypass_media=true"/> before
>>> bridging.
>>>
>>> 3. Neither of the above in the extension.
>>>
>>>  Only 2 with "bypass-media=true" gets the audio across endpoints.
>>>
>>> Help :-)
>>>
>>>
>>>  -----Original Message-----
>>> From: mszlazak at aol.com
>>> To: freeswitch-users at lists.freeswitch.org
>>>  Sent: Fri, 19 Dec 2008 11:30 am
>>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>>> Media work?
>>>
>>>    With the firewall ON or OFF the problem still remains.
>>>
>>> I've tried 3 different set-ups in a dial plan extension.
>>>
>>> 1. With  only <action application="set" data="proxy_media=true"/> before
>>> bridging.
>>>
>>> 2. With only  <action application="set" data="bypass_media=true"/> before
>>> bridging.
>>>
>>> 3. Neither of the above in the extension.
>>>
>>> Only 2 with "proxy-media=true" gets the audio across endpoints.
>>>
>>> Help :-)
>>>
>>>
>>>
>>>
>>>
>>>  0A
>>>
>>>
>>>  -----Original Message-----
>>> From: Michael Jerris <mike at jerris.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Fri, 19 Dec 2008 7:49 am
>>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy
>>> Media work?
>>>
>>>  It gives me the impression there is something wrong with your firewall
>>> running on the box.
>>>  Mike
>>>
>>>   On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>>>
>>> I find it strange that I can have to endpoints get audio went using
>>> bypass media mode but the audio fails to go across endpoints if I use proxy
>>> media mode.
>>> I'm trying to pass audio "internally" on the same machine between
>>> endpoints and have be advis ed that a reason the audio may fail to be passed
>>> is because there is some RTP timing and IP address/port issues.
>>> However, FS has no problem "connecting" ports if i change the mode to
>>> bypass media. This gives me the impression that something is wrong with FS
>>> proxy media mode.
>>> Any comments?
>>>
>>> ------------------------------
>>> Listen to 350+ music, sports, & news radio stations ? including songs for
>>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>>
>>>  _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch. org <http://www.freeswitch.org>
>>>
>>>
>>>  =
>>>
>>> _______________________________________________
>>>
>>>
>>>
>>>
>>> Freeswitch-users mailing list
>>>
>>>
>>>
>>> Freeswitch-users at lists.freeswitch.org
>>>
>>>
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>>
>>>
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>>
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> ------------------------------
>>> Listen to 350+ music, sports, & news radio stations ? including songs for
>>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>>
>>>
>>> _______________________________________________
>>>
>>>
>>>
>>>
>>> Freeswitch-users mailing list
>>>
>>> Freeswitch-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> ------------------------------
>>> Listen to 350+ music, sports, & news radio stations ? including songs for
>>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<http://lists.freeswitch.org/mailman/listinfo++/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>>
>> Freeswitch-users mailing list
>> Freeswitch-users at lists
>> .freeswitch.org <Freeswitch-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> Listen to 350+ music, sports, & news radio stations ? including songs for
>> the holidays ? FREE while you browse. Start Listening Now<http://toolbar.aol.com/aolradio/download.html?ncid=emlweusdown00000013>!
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Dec 22 08:02:02 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 10:02:02 -0600
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.umklwrrxatcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
Message-ID: <191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>

are you doing the trace on the FS box too?
it says it's established RTP and bridging.

NO audio is 9.8/10 times a firewall issue.

typing in a message is not the right way to call someone on jingle.
That's the point.  In component mode you add the sip ext to your buddy list
and call them the normal way.  This has nothing to do with your audio issue
though so it's
not a big deal.

On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> There are absolutely no UDP packets going trough like when doing a call
>  from gtalk to gtalk.
>
> You mean this (component mode):
>
> http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F
> Is there more documentation that this?
>
> All I would like to do is to initiate a call between SIP telephone and
> gtalk user who typed in the message.
>
> Thank you!
>
>
> On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>
> > Your log shows rtp streams being allocated.
> > did you look at at the packets on the wire with a packet capture program?
> >
> > You are better off using proper jingle and component mode.  What you are
> > describing sounds like
> > a workaround to avoid doing it right.
> >
> >
> >
> > On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:
> >
> >> I modified mod_dingaling.c so I can intercept google talk chat messages
> >> via socket - nothing fancy.
> >> Then I wrote a java app that connects to freeswitch socket and in case
> >> of
> >> a proper message (trigger) it sends a command to freeswitch, in my case:
> >> api originate sofia/default/1001 at 10.99.8.221
> >> &bridge(dingaling/gmail.com/my_mail at gmail.com)
> >>
> >> Dingaling is logged in as another user which I have added as buddy, chat
> >> messages go trough, however when a call is started
> >> between SIP and Gtalk client, we cannot hear each other at all.
> >> Using freeswitch revision: 10866
> >>
> >> Here is the log:
> >> http://pastebin.com/m1eba2cb8
> >>
> >> What can be the problem? First I thought it is because running sip
> >> client
> >> + gtalk and freeswitch on one host, but then I
> >> moved SIP phone and Gtalk to 2 different workstations, using the third
> >> only for freeswitch. Also calls from "call" example program
> >>  from google lib works fine with same setup - something must be
> >> problematic
> >> with freeswitch, however cannot see what.
> >>
> >> Thank you!
> >>
> >> --
> >> kriko
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
>
>
>
> --
> Porn - the reason you need a new hard drive.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristjan.ugrin at gmail.com  Mon Dec 22 08:19:50 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Mon, 22 Dec 2008 17:19:50 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
Message-ID: <op.umknnbetatcbto@linux.site>

But what I would like to achieve is something different (quite similar).
You type in a message like "call 1001 at 10.99.8.20" and you it would call a  
SIP buddy with any local number.

In component mode you need to add a buddy everytime for a different sip  
nr.?
Which would mean a lot of numbers if you would like to call more than one  
sip nr. in a lan.

As for the first issue, there are RTP packets traveling on FS, but never  
reach destination after they leave our internal network.
Do they have to go outside lan even when the call is placed in a lan  
between gtalk and SIP?
Gtalk to gtalk is no problem on same machines...


On Mon, 22 Dec 2008 17:02:02 +0100, Anthony Minessale  
<anthony.minessale at gmail.com> wrote:

> are you doing the trace on the FS box too?
> it says it's established RTP and bridging.
>
> NO audio is 9.8/10 times a firewall issue.
>
> typing in a message is not the right way to call someone on jingle.
> That's the point.  In component mode you add the sip ext to your buddy  
> list
> and call them the normal way.  This has nothing to do with your audio  
> issue
> though so it's
> not a big deal.
>
> On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> There are absolutely no UDP packets going trough like when doing a call
>>  from gtalk to gtalk.
>>
>> You mean this (component mode):
>>
>> http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F
>> Is there more documentation that this?
>>
>> All I would like to do is to initiate a call between SIP telephone and
>> gtalk user who typed in the message.
>>
>> Thank you!
>>
>>
>> On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>
>> > Your log shows rtp streams being allocated.
>> > did you look at at the packets on the wire with a packet capture  
>> program?
>> >
>> > You are better off using proper jingle and component mode.  What you  
>> are
>> > describing sounds like
>> > a workaround to avoid doing it right.
>> >
>> >
>> >
>> > On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com>  
>> wrote:
>> >
>> >> I modified mod_dingaling.c so I can intercept google talk chat  
>> messages
>> >> via socket - nothing fancy.
>> >> Then I wrote a java app that connects to freeswitch socket and in  
>> case
>> >> of
>> >> a proper message (trigger) it sends a command to freeswitch, in my  
>> case:
>> >> api originate sofia/default/1001 at 10.99.8.221
>> >> &bridge(dingaling/gmail.com/my_mail at gmail.com)
>> >>
>> >> Dingaling is logged in as another user which I have added as buddy,  
>> chat
>> >> messages go trough, however when a call is started
>> >> between SIP and Gtalk client, we cannot hear each other at all.
>> >> Using freeswitch revision: 10866
>> >>
>> >> Here is the log:
>> >> http://pastebin.com/m1eba2cb8
>> >>
>> >> What can be the problem? First I thought it is because running sip
>> >> client
>> >> + gtalk and freeswitch on one host, but then I
>> >> moved SIP phone and Gtalk to 2 different workstations, using the  
>> third
>> >> only for freeswitch. Also calls from "call" example program
>> >>  from google lib works fine with same setup - something must be
>> >> problematic
>> >> with freeswitch, however cannot see what.
>> >>
>> >> Thank you!
>> >>
>> >> --
>> >> kriko
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>>
>>
>>
>> --
>> Porn - the reason you need a new hard drive.
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>



-- 



From msc at freeswitch.org  Mon Dec 22 08:36:29 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 08:36:29 -0800
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <1229954080.9989.41.camel@worek.man.szczecin.pl>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>
Message-ID: <87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>

Seweryn,

Which phone(s) are you using? FS does BLF very well with Snom, Grandstream,
and Linksys. Also, the dialing scenario you mention is actually very easily
handled with FreeSWITCH. The devs are very clever and they set up a dialing
syntax mechanism that allows one to do all sorts of unique and even exotic
dialing setups.

Do you have a spare Linux machine and a few phones that you can do some
testing with? That would be the ideal way to get started quickly. If you do
have a Linux machine then the quickest way to get FS running is to do this:

http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install

Just remember one thing: FreeSWITCH is quite different from Asterisk, so
there is a bit of a learning curve, but it's totally worth it. :)

-MC

On Mon, Dec 22, 2008 at 5:54 AM, Seweryn Niemiec <ser at man.szczecin.pl>wrote:

> Hi,
>
> I'm currently evaluating FreeSWITCH to see if I can migrate to it from
> Asterisk (which gives me now all functions I need, but has some problems
> with predictability). I need only one atypical functionality:
> 1. we have one main extension, which clients call. lets name it 5555
> 2. when someone calls 5555, then immediately rings only one
>   phone (1001) and 15 seconds later 3 more phones (1002-1004). phone
>   1001 should be dialled once per call to 5555
> 3. all phones (1000-1010) have BLF monitoring 5555, so anyone can
>   pick it up when someone calls it
>
> ad 3. it is acceptable that all phones monitor 1001 instead of 5555 to
> pick up connection to 5555
>
> After reading and googling about possible implementation in FS for few
> hours I couldn't find anything useful.
>
> Could you tell if it is possible to implement in FS without artificial
> limbs (i dunno if it's best English word for what I mean :) like I had
> to do in Asterisk?
>
> greetings,
> Seweryn
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Dec 22 08:44:22 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 10:44:22 -0600
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>
	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
Message-ID: <BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>

The default config demo's this.. its using the db app to store the  
UUID and the intercept app to pickup the call.

/b

On Dec 22, 2008, at 10:36 AM, Michael Collins wrote:

> After reading and googling about possible implementation in FS for few
> hours I couldn't find anything useful.




From kawarod at laposte.net  Mon Dec 22 07:55:03 2008
From: kawarod at laposte.net (rod)
Date: Mon, 22 Dec 2008 19:55:03 +0400
Subject: [Freeswitch-users] SIP Headers and use of "" in the header
Message-ID: <494FB857.9090003@laposte.net>

Dear All,

I've been playing with the freeswitch options for one month now, and 
I've been able to use it with kamailio working as a registrar.

What I'd like to do is to add a diversion header using the following 
action in the dialplan:
    <action application="export" 
data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="unconditional""/>

Please note, that I'd like to put the word "unconditional" between 
quotes, this is to comply with the SIP gateway to which I'm sending trafic.

But I've been unable to set an escape character to use theses quotes, 
cause as you may understand, without escape character FS will consider 
this instruction instead:

<action application="export" 
data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="

cause the quote after reason= is considered as a closing quote for data=".

Is there a way to achieve this.

Thanks.
rod



From Laurent.Fabre at kirranet.com  Mon Dec 22 08:20:32 2008
From: Laurent.Fabre at kirranet.com (Laurent Fabre)
Date: Mon, 22 Dec 2008 17:20:32 +0100
Subject: [Freeswitch-users] Freeswitch/Sofia configuration problem
Message-ID: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>


Hello,

I've been trying to figure out for a few days why my freeswitch instance suddenly become insensitive to SIP packets without any warning.

What usually happen is the following :

1) start just fine in foreground mode and no errors
2) wait anywhere between 2 seconds and 20 minutes 
3) Sofia suddenly decide to reload everything for some reason
4) Sofia start processing SIP packets
5) work for an hour or so
6) Sofia suddenly decide to reload everything for some reason
7) become unresponsive again
8) goto 2

Both interfaces have public IP addresses assigned in a static manner (no DHCP).

I can see the SIP UDP & TCP requests comming from the phones on several sites on the wire.
The SIP TCP requests get RST in reply which is mean :(

There was a point in my setup where it would not happen but since I'm new to freeswitch I'm having an hard time backtracking.

I was wondering if iproute/tc and iptables were the culprits but I flushed everything (even rebooted without loading the rules) and it still doesn't work.

I thought some database was corrupt so I shutdown'd freeswitch and delete his db folder, no effect.

My server runs Debian 4.0etch for amd64, built freeswitch from SVN trunk.

Any pointers, help, cure against headaches would be great :)

Regards,

Laurent




From mike at jerris.com  Mon Dec 22 09:10:33 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Dec 2008 12:10:33 -0500
Subject: [Freeswitch-users] SIP Headers and use of "" in the header
In-Reply-To: <494FB857.9090003@laposte.net>
References: <494FB857.9090003@laposte.net>
Message-ID: <2BC8260A-5118-48E4-9F5B-60F5547A59D6@jerris.com>

You should be able to do this with xml CDATA syntax.

Mike

On Dec 22, 2008, at 10:55 AM, rod wrote:

> Dear All,
>
> I've been playing with the freeswitch options for one month now, and
> I've been able to use it with kamailio working as a registrar.
>
> What I'd like to do is to add a diversion header using the following
> action in the dialplan:
>    <action application="export"
> data
> ="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="unconditional""/>
>
> Please note, that I'd like to put the word "unconditional" between
> quotes, this is to comply with the SIP gateway to which I'm sending  
> trafic.
>
> But I've been unable to set an escape character to use theses quotes,
> cause as you may understand, without escape character FS will consider
> this instruction instead:
>
> <action application="export"
> data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="
>
> cause the quote after reason= is considered as a closing quote for  
> data=".




From brian at freeswitch.org  Mon Dec 22 09:11:26 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 11:11:26 -0600
Subject: [Freeswitch-users] SIP Headers and use of "" in the header
In-Reply-To: <494FB857.9090003@laposte.net>
References: <494FB857.9090003@laposte.net>
Message-ID: <DE122F9B-6648-4549-96FC-037BFF81BB98@freeswitch.org>

<action application="export"><![CDATA[sip_h_Diversion=<123456789 at 10.10.10.254 
 >;reason="unconditional"]]></action>

/b

On Dec 22, 2008, at 9:55 AM, rod wrote:

> Dear All,
>
> I've been playing with the freeswitch options for one month now, and
> I've been able to use it with kamailio working as a registrar.
>
> What I'd like to do is to add a diversion header using the following
> action in the dialplan:
>    <action application="export"
> data 
> ="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="unconditional""/>
>
> Please note, that I'd like to put the word "unconditional" between
> quotes, this is to comply with the SIP gateway to which I'm sending  
> trafic.
>
> But I've been unable to set an escape character to use theses quotes,
> cause as you may understand, without escape character FS will consider
> this instruction instead:
>
> <action application="export"
> data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="
>
> cause the quote after reason= is considered as a closing quote for  
> data=".
>
> Is there a way to achieve this.
>
> Thanks.
> rod
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Mon Dec 22 09:30:46 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 11:30:46 -0600
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.umknnbetatcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
Message-ID: <191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>

if you see them leave FS and never reach dest.
It implies a firewall somewhere in between is blocking them.


On Mon, Dec 22, 2008 at 10:19 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> But what I would like to achieve is something different (quite similar).
> You type in a message like "call 1001 at 10.99.8.20" and you it would call a
> SIP buddy with any local number.
>
> In component mode you need to add a buddy everytime for a different sip
> nr.?
> Which would mean a lot of numbers if you would like to call more than one
> sip nr. in a lan.
>
> As for the first issue, there are RTP packets traveling on FS, but never
> reach destination after they leave our internal network.
> Do they have to go outside lan even when the call is placed in a lan
> between gtalk and SIP?
> Gtalk to gtalk is no problem on same machines...
>
>
> On Mon, 22 Dec 2008 17:02:02 +0100, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>
> > are you doing the trace on the FS box too?
> > it says it's established RTP and bridging.
> >
> > NO audio is 9.8/10 times a firewall issue.
> >
> > typing in a message is not the right way to call someone on jingle.
> > That's the point.  In component mode you add the sip ext to your buddy
> > list
> > and call them the normal way.  This has nothing to do with your audio
> > issue
> > though so it's
> > not a big deal.
> >
> > On Mon, Dec 22, 2008 at 9:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:
> >
> >> There are absolutely no UDP packets going trough like when doing a call
> >>  from gtalk to gtalk.
> >>
> >> You mean this (component mode):
> >>
> >>
> http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F
> >> Is there more documentation that this?
> >>
> >> All I would like to do is to initiate a call between SIP telephone and
> >> gtalk user who typed in the message.
> >>
> >> Thank you!
> >>
> >>
> >> On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>
> >> > Your log shows rtp streams being allocated.
> >> > did you look at at the packets on the wire with a packet capture
> >> program?
> >> >
> >> > You are better off using proper jingle and component mode.  What you
> >> are
> >> > describing sounds like
> >> > a workaround to avoid doing it right.
> >> >
> >> >
> >> >
> >> > On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com>
> >> wrote:
> >> >
> >> >> I modified mod_dingaling.c so I can intercept google talk chat
> >> messages
> >> >> via socket - nothing fancy.
> >> >> Then I wrote a java app that connects to freeswitch socket and in
> >> case
> >> >> of
> >> >> a proper message (trigger) it sends a command to freeswitch, in my
> >> case:
> >> >> api originate sofia/default/1001 at 10.99.8.221
> >> >> &bridge(dingaling/gmail.com/my_mail at gmail.com)
> >> >>
> >> >> Dingaling is logged in as another user which I have added as buddy,
> >> chat
> >> >> messages go trough, however when a call is started
> >> >> between SIP and Gtalk client, we cannot hear each other at all.
> >> >> Using freeswitch revision: 10866
> >> >>
> >> >> Here is the log:
> >> >> http://pastebin.com/m1eba2cb8
> >> >>
> >> >> What can be the problem? First I thought it is because running sip
> >> >> client
> >> >> + gtalk and freeswitch on one host, but then I
> >> >> moved SIP phone and Gtalk to 2 different workstations, using the
> >> third
> >> >> only for freeswitch. Also calls from "call" example program
> >> >>  from google lib works fine with same setup - something must be
> >> >> problematic
> >> >> with freeswitch, however cannot see what.
> >> >>
> >> >> Thank you!
> >> >>
> >> >> --
> >> >> kriko
> >> >>
> >> >> _______________________________________________
> >> >> Freeswitch-users mailing list
> >> >> Freeswitch-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Porn - the reason you need a new hard drive.
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
>
>
>
> --
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mszlazak at aol.com  Mon Dec 22 09:53:04 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 22 Dec 2008 12:53:04 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com><B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com><8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com><8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com><191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com><8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com><191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
	<8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
Message-ID: <8CB327545F492AF-928-221@WEBMAIL-MA02.sysops.aol.com>


 Hi Andy and Anthony.

Thanks Anthony for elaborating more and I'll attempt using another IP on the same box as well. 

Also, Prophecy support has asked me first to put one application on a separate box and then get some wireshark data so I'll attempt that also.

Andy, I didn't want to bother you given all those things you had to deal with. Welcome back.

I explored the VMware idea before but was warned that it would not work well with an ASR. This advice came from the Trixbox forums, LumenVox, FreeSwitch and Voxeo.

I understand that what I'm doing goes against the grain (i.e. voip) but frankly my target market really doesn't want anything to do with voip or even internet connectivity from their businesses. Plus there are other issues.

I'll let you know how it goes.

Happy holidays.

Mark.


 

-----Original Message-----
From: Andrew Gilbert <gilbertandrew at me.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 22 Dec 2008 7:49 am
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?









Mark,



Sorry I haven't had much time to help with this either.




But Anthony is offering good advice here.?You are either going to have to work out what is going on at SIP/SDP/RTP level through logs and wireshark, or opt for a separate ip space. Another option (besides virtual ips) is VMWare or VirtualBox, although VMWare is probably easier to setup and bridge naturally to your host.




Vm's are just so easy anymore and it definite
ly seems like you are going against the grain right now.




Also - realizing you got here because of the need for ASR. I do have the Lumenvox license, and I was able to compile the module out of SVN. I have not tested anything yet. If things go well I should have some time after the 25th for this. My goal would be to get pizza or something akin to work.




Andy







On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:


I don't really know what your problem is.? I just saw you ask 3 times for help and tried to offer a suggestion.
if you start FS with TPORT_LOG=1 you can see all the sip messages in the console and you could 
also run wireshark to look at a packet capture.
 
If you use the same IP for media on the same box for 3 programs at once you may end up with 2 applictions choosing the same media port etc.

It's just a good practice to run every voip program on it's own IP.
 




On Mon, Dec 22, 2008 at 12:44 AM,  <mszlazak at aol.com> wrote:

 
 Hi Anthony,
 
 I actually suggested adding IP's to a Voxeo-Prophecy support person before but they thought that could be problematic. I went along with the earlier warning but now you have suggested it again. What makes everything on the same box tricky? 
 
 Also, the thing that surprises me a bit is that bypass-media works but proxy-media or the default doesn't. 
 
 Would you be kind enough to elaborate.
 
 Thanks. Mark.
 
 

 
 
 
 
20
 
 -----Original Message-----
 From: Anthony Minessale <anthony.minessale at gmail.com>
 To: freeswitch-users at lists.freeswitch.org





 Sent: Sun, 21 Dec 2008 2:49 pm
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 Try adding more ip to your box and give each thing it's own dedicated virtual IP.
 Doing everything on the same box can be tricky.
 
 
 
On Sat, Dec 20, 2008 at 2:17 AM,  <mszlazak at aol.com> wrote:
 
 
With the firewall ON or OFF the problem still remains.
 
 I've tried 3 different set-ups in a dial plan extension.
 
 1. With? only <action application="set" data="proxy_media=true"/> before bridging.
 ??? ??? ??? 
 2. With only? <action application="set" data="bypass_media=true"/> before bridging.
 
 3. Neither of the above in the extension.
 
 
 Only 2 with "bypass-media=true" gets the audio across endpoints.
 
 Help :-)
  
 
 
 
 
 
 
 -----Original Message-----
 From: mszlazak at aol.com
 To: freeswitch-users at lists.freeswitch.org
 
 
 Sent: Fri, 19 Dec 2008 11:30 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 
 
 
?With the firewall ON or OFF the problem still remains.
 
 I've tried 3 different set-ups in a dial plan extension.
 
 1. With? only <action application="set" data="proxy_media=true"/> before bridging.
 ??? ??? ??? 
 2. With only? <action app
lication="set" data="bypass_media=true"/> before bridging.
 
 3. Neither of the above in the extension.
 
 Only 2 with "proxy-media=true" gets the audio across endpoints.
 
 Help :-)
 
 
 ? 
 
 
 
 
 0A 
 

 
 
 
 
 
 
 
 -----Original Message-----
 From: Michael Jerris <mike at jerris.com>
 To: freeswitch-users at lists.freeswitch.org
 Sent: Fri, 19 Dec 2008 7:49 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 It gives me the impression there is something wrong with your firewall running on the box. 

 
 
Mike
 

 
 
 
 
On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
 
 
I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.
  I'm trying to pass audio "internally" on the same machine between endpoints and have be advis ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 
  However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.
  Any comments?? 
   

 

Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
  _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lis
ts.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch. org
 

 
 
 = 
  
 
_______________________________________________







Freeswitch-users mailing list







Freeswitch-users at lists.freeswitch.org







http://lists.freeswitch.org/mailman/listinfo/freeswitch-users







UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users







http://www.freeswitch.org








 
  

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 
 

 
   
 
_______________________________________________






Freeswitch-users mailing list



Freeswitch-users at lists.freeswitch.org



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



http://www.freeswitch.org




 
   

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 

 
 

 -- 
 Anthony Minessale II
 
 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 
 AIM:
 anthm
 MSN:anthony_minessale at hotmail.com
 GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
 IRC: irc.freenode.net #freeswitch
 
 FreeSWITCH Developer Conference
 sip:888 at conference.freeswitch.org
 iax:guest at conference.freeswitch.org/888
 googletalk:conf+888 at conference.freeswitch.org
 pstn:213-799-1400
 
 
 
_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists
.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org


 
 



 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 


_______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 





-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
 
AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
 IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
 googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400
 _______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




=


 





_______
________________________________________
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From anthony.minessale at gmail.com  Mon Dec 22 09:59:15 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Dec 2008 11:59:15 -0600
Subject: [Freeswitch-users] Mod Fax: Error, problems and questions...
In-Reply-To: <5e414ed0812190933j772292bdw32bbb7213c6b6591@mail.gmail.com>
References: <5e414ed0812041145gaa63014ga3c1c678511cc31d@mail.gmail.com>
	<49387485.9000303@coppice.org>
	<5e414ed0812050254s711f0b66y20108d9cc2d96379@mail.gmail.com>
	<3DE12536-6B7F-4F05-B4CE-2A30A93BB3F9@jerris.com>
	<5e414ed0812190844w7bbfc4d5sc7a81eb2fadad091@mail.gmail.com>
	<191c3a030812190900l4037ead8m4fefa87fb55fb82d@mail.gmail.com>
	<5e414ed0812190933j772292bdw32bbb7213c6b6591@mail.gmail.com>
Message-ID: <191c3a030812220959y1c4a754cu2c494e28f9324514@mail.gmail.com>

no difference.


On Fri, Dec 19, 2008 at 11:33 AM, Dennis <odermann at googlemail.com> wrote:

> ahh, just a second. it seems that i did not realize a small
> missunderstanding in you answer.
>
> i do not want to SEND a fax, i just want to RECEIVE a fax. so the fax
> comes in at out carrier and the rest is sent over about 1m of cat6 to
> our fs server.
>
> is there a difference or does it not matter, if we want to receive or
> send a fax?
>
>
>
>
>
> 2008/12/19 Anthony Minessale <anthony.minessale at gmail.com>:
> > You don't know where the audio goes after that switch in the same room
> until
> > it gets to the guy
> > with the fax machine.
> >
> > No it will not be improved by Christmas.  Not a chance.
> >
> > Yes it will probably be much more reliable once it can do T38.
> >
> > Be happy with what you have for the holiday season.
> >
> >
> >
> > On Fri, Dec 19, 2008 at 10:44 AM, Dennis <odermann at googlemail.com>
> wrote:
> >>
> >> it's me again about mod fax... it is short before christmas and my
> >> whish is, to get mod fax working quite reliable. is this possible
> >> under optimal conditions?
> >>
> >> all our tests lead by far to more failed faxes than received faxes. i
> >> really like the fax feature and would like to see it beeing usable.
> >>
> >> is it just pure luck, if a fax was received or are there some
> >> conditions out there, which could help beeing mod fax reliable?
> >> second question: what about t38? will it come? is there chance, that
> >> it will come? where are the difficulties with mod fax?
> >>
> >> our fs servers are standing directly beside the sip switch of our
> >> carrier. from the carriers switch, there is a 50 cm long cat6 cable
> >> going into our cisco-switch. from the cisco switch there are 50 cm
> >> long cat6 cables going into our fs servers.
> >> i doubt, that there can be a signifant packet loss.
> >> are there some settings, we could try out or is the faxing stuff just
> >> unusable, till t38 is supported?
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From wiltingtree at gmail.com  Mon Dec 22 09:34:19 2008
From: wiltingtree at gmail.com (Adam Wilt)
Date: Mon, 22 Dec 2008 12:34:19 -0500
Subject: [Freeswitch-users] Phone lines ring busy after awhile
Message-ID: <eb7797d70812220934v449243fhf6508bd892d92a0f@mail.gmail.com>

Hello.  I have an installation of FreeSwitch runnnig, and I wrote a Python
script to answer an inbound call and play an IVR.  It works fine for awhile,
but eventually it stops answering the phone line and the phone rings busy.
When I look in the FreeSwitch logs I don't see anything unusual. But I don't
think it's a problem with the VOIP phone service because when I restart
FreeSwitch the problem goes away.  Any ideas?
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From ser at man.szczecin.pl  Mon Dec 22 12:05:53 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Mon, 22 Dec 2008 21:05:53 +0100
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>
	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
Message-ID: <494FF321.1000207@man.szczecin.pl>

On Mon, Dec 22, 2008 at 5:36 PM, Michael Collins <msc at freeswitch.org> wrote:
> Which phone(s) are you using? FS does BLF very well with Snom, 
> Grandstream,

I have Grandstreams and I have already tested BLF and "standard"
pickup on FS. It works great on default configuration.

> and Linksys. Also, the dialing scenario you mention is actually very
> easily  handled with FreeSWITCH. The devs are very clever and they 
> set up a dialing syntax mechanism that allows one to do all sorts of
> unique and even exotic dialing setups.

are "bridge" and "sleep" actions + some FS's magic all what I need?

> Do you have a spare Linux machine and a few phones that you can do 
> some testing with? That would be the ideal way to get started 
> quickly. If you do
> have a Linux machine then the quickest way to get FS running is to do

First I tried to compile FS on my workstation (Ubuntu) but it has
failed due to too new libtool. But on latest Debian it was OK and I
have FS console under my fingers.

> Just remember one thing: FreeSWITCH is quite different from Asterisk,
>  so there is a bit of a learning curve, but it's totally worth it. :)

BTW: it would be nice to have some notes about reloading configs in
"getting started docs" on the wiki.

--
Best regards
Seweryn Niemiec



From ser at man.szczecin.pl  Mon Dec 22 12:14:26 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Mon, 22 Dec 2008 21:14:26 +0100
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
	<BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>
Message-ID: <494FF522.8030307@man.szczecin.pl>

On Mon, Dec 22, 2008 at 5:44 PM, Brian West <brian at freeswitch.org> wrote:
> The default config demo's this.. its using the db app to store the
 > UUID and the intercept app to pickup the call.

OK, first things first, can I monitor 5555 with BLF? Because if not,
then we get to the same point like in Asterisk, where I do:
- redirect 5555 to 1555 on upper tier (PSTN)
- "terminate" 5555 on the phone with 1001 extension
- group dial 5555 and 1002-1004 with delay when 1555 is called
Now 5555 can be monitored with BLF.

The second thing (important only when using above trick) is: when there
is a call to 1555 and phone 5555 is ringing, can I pick it up by **5555?

Because if not, then again we get to the same problem like in Asterisk. In
Asterisk when rings the phone with 5555 extension you have to pickup
1555 not 5555. So on BLF you monitor 5555 but to pickup you dial **1555.
This is quite stupid, users can't have two buttons to service one
extension (one for BLF and one for pickup). To get it working
transparently for endusers on Asterisk I had to hack and slash
1555 and **XXXX extensions (ugly global variables involved).

--
Best regards
Seweryn Niemiec



From msc at freeswitch.org  Mon Dec 22 12:16:08 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 12:16:08 -0800
Subject: [Freeswitch-users] Phone lines ring busy after awhile
In-Reply-To: <eb7797d70812220934v449243fhf6508bd892d92a0f@mail.gmail.com>
References: <eb7797d70812220934v449243fhf6508bd892d92a0f@mail.gmail.com>
Message-ID: <87f2f3b90812221216i42cbaeedx52cee1863e68667a@mail.gmail.com>

what operating system and which revision of FreeSWITCH?
-MC

On Mon, Dec 22, 2008 at 9:34 AM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hello.  I have an installation of FreeSwitch runnnig, and I wrote a Python
> script to answer an inbound call and play an IVR.  It works fine for awhile,
> but eventually it stops answering the phone line and the phone rings busy.
> When I look in the FreeSwitch logs I don't see anything unusual. But I don't
> think it's a problem with the VOIP phone service because when I restart
> FreeSwitch the problem goes away.  Any ideas?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Dec 22 12:18:23 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 14:18:23 -0600
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <494FF522.8030307@man.szczecin.pl>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>
	<BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>
	<494FF522.8030307@man.szczecin.pl>
Message-ID: <6808A7E0-C0B2-4A8C-9D29-B045ABC37938@freeswitch.org>

The tip I can give you here is you have to set the  
"presence_id=5555 at domain" on a session so 1. the events are fired for  
that... and 2. you actually get the BLF for 5555  ;)  The presence ID  
will allow you to set arbitrary things you can then subscribe to on  
the phone.

/b

On Dec 22, 2008, at 2:14 PM, Seweryn Niemiec wrote:

> On Mon, Dec 22, 2008 at 5:44 PM, Brian West <brian at freeswitch.org>  
> wrote:
>> The default config demo's this.. its using the db app to store the
>> UUID and the intercept app to pickup the call.
>
> OK, first things first, can I monitor 5555 with BLF? Because if not,
> then we get to the same point like in Asterisk, where I do:
> - redirect 5555 to 1555 on upper tier (PSTN)
> - "terminate" 5555 on the phone with 1001 extension
> - group dial 5555 and 1002-1004 with delay when 1555 is called
> Now 5555 can be monitored with BLF.
>
> The second thing (important only when using above trick) is: when  
> there
> is a call to 1555 and phone 5555 is ringing, can I pick it up by  
> **5555?
>
> Because if not, then again we get to the same problem like in  
> Asterisk. In
> Asterisk when rings the phone with 5555 extension you have to pickup
> 1555 not 5555. So on BLF you monitor 5555 but to pickup you dial  
> **1555.
> This is quite stupid, users can't have two buttons to service one
> extension (one for BLF and one for pickup). To get it working
> transparently for endusers on Asterisk I had to hack and slash
> 1555 and **XXXX extensions (ugly global variables involved).
>
> --
> Best regards
> Seweryn Niemiec
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From wiltingtree at gmail.com  Mon Dec 22 12:27:08 2008
From: wiltingtree at gmail.com (Adam Wilt)
Date: Mon, 22 Dec 2008 15:27:08 -0500
Subject: [Freeswitch-users] Phone lines ring busy after awhile
Message-ID: <eb7797d70812221227o65eb07a4y141be8e5808989e1@mail.gmail.com>

 Sorry,  I'm using freeswitch-1.0.latest.tar.gz on Fedora.


>
> Message: 5
> Date: Mon, 22 Dec 2008 12:16:08 -0800
> From: "Michael Collins" <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] Phone lines ring busy after awhile
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>        <87f2f3b90812221216i42cbaeedx52cee1863e68667a at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> what operating system and which revision of FreeSWITCH?
> -MC
>
> On Mon, Dec 22, 2008 at 9:34 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
>
> > Hello.  I have an installation of FreeSwitch runnnig, and I wrote a
> Python
> > script to answer an inbound call and play an IVR.  It works fine for
> awhile,
> > but eventually it stops answering the phone line and the phone rings
> busy.
> > When I look in the FreeSwitch logs I don't see anything unusual. But I
> don't
> > think it's a problem with the VOIP phone service because when I restart
> > FreeSwitch the problem goes away.  Any ideas?
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
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From ser at man.szczecin.pl  Mon Dec 22 12:28:31 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Mon, 22 Dec 2008 21:28:31 +0100
Subject: [Freeswitch-users] group call with BLF and pickup
In-Reply-To: <6808A7E0-C0B2-4A8C-9D29-B045ABC37938@freeswitch.org>
References: <1229954080.9989.41.camel@worek.man.szczecin.pl>	<87f2f3b90812220836p70e3e443gd02bb3f800cfca2@mail.gmail.com>	<BDBBE10A-E709-41DB-84FB-EA257A105E45@freeswitch.org>	<494FF522.8030307@man.szczecin.pl>
	<6808A7E0-C0B2-4A8C-9D29-B045ABC37938@freeswitch.org>
Message-ID: <494FF86F.1030000@man.szczecin.pl>

Brian West wrote:
> The tip I can give you here is you have to set the  
> "presence_id=5555 at domain" on a session so 1. the events are fired for  
> that... and 2. you actually get the BLF for 5555  ;)  The presence ID  
> will allow you to set arbitrary things you can then subscribe to on  
> the phone.

that sounds cool. i'll start experiments tomorrow. thx for very fast reply.

-- 
greetings,
seweryn



From msc at freeswitch.org  Mon Dec 22 12:41:23 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 12:41:23 -0800
Subject: [Freeswitch-users] Phone lines ring busy after awhile
In-Reply-To: <eb7797d70812221227o65eb07a4y141be8e5808989e1@mail.gmail.com>
References: <eb7797d70812221227o65eb07a4y141be8e5808989e1@mail.gmail.com>
Message-ID: <87f2f3b90812221241w7dece8e2w54019ffeec0bead3@mail.gmail.com>

Hmmm... I think that might be a problem. I think that file hasn't been
getting updates - it looks way old on files.freeswitch.org.

Would you mind doing a few things to get your system on the latest? First,
mv /usr/local/freeswitch /usr/local/freeswitch.old. That will preserve your
existing config for future reference. Then do a new svn checkout and build
the latest, which is much improved. I recommend this process:

http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install

After you get the fresh install done then add back in your customizations
and try your script. Let us know how it goes.

-MC

On Mon, Dec 22, 2008 at 12:27 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Sorry,  I'm using freeswitch-1.0.latest.tar.gz on Fedora.
>
>
>>
>> Message: 5
>> Date: Mon, 22 Dec 2008 12:16:08 -0800
>> From: "Michael Collins" <msc at freeswitch.org>
>> Subject: Re: [Freeswitch-users] Phone lines ring busy after awhile
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID:
>>        <87f2f3b90812221216i42cbaeedx52cee1863e68667a at mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>>
>> what operating system and which revision of FreeSWITCH?
>> -MC
>>
>> On Mon, Dec 22, 2008 at 9:34 AM, Adam Wilt <wiltingtree at gmail.com> wrote:
>>
>> > Hello.  I have an installation of FreeSwitch runnnig, and I wrote a
>> Python
>> > script to answer an inbound call and play an IVR.  It works fine for
>> awhile,
>> > but eventually it stops answering the phone line and the phone rings
>> busy.
>> > When I look in the FreeSwitch logs I don't see anything unusual. But I
>> don't
>> > think it's a problem with the VOIP phone service because when I restart
>> > FreeSwitch the problem goes away.  Any ideas?
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Mon Dec 22 12:48:05 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 22 Dec 2008 15:48:05 -0500
Subject: [Freeswitch-users] Phone lines ring busy after awhile
In-Reply-To: <87f2f3b90812221241w7dece8e2w54019ffeec0bead3@mail.gmail.com>
References: <eb7797d70812221227o65eb07a4y141be8e5808989e1@mail.gmail.com>
	<87f2f3b90812221241w7dece8e2w54019ffeec0bead3@mail.gmail.com>
Message-ID: <A65C0A69-0771-46A8-B71F-0D1E65240031@jerris.com>

1.0.latest points to the last 1.0 release which was 1.0.1.

Mike

On Dec 22, 2008, at 3:41 PM, Michael Collins wrote:

> Hmmm... I think that might be a problem. I think that file hasn't  
> been getting updates - it looks way old on files.freeswitch.org.
>
> Would you mind doing a few things to get your system on the latest?  
> First, mv /usr/local/freeswitch /usr/local/freeswitch.old. That will  
> preserve your existing config for future reference. Then do a new  
> svn checkout and build the latest, which is much improved. I  
> recommend this process:
>
> http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
>
> After you get the fresh install done then add back in your  
> customizations and try your script. Let us know how it goes.
>
> -MC
>
> On Mon, Dec 22, 2008 at 12:27 PM, Adam Wilt <wiltingtree at gmail.com>  
> wrote:
> Sorry,  I'm using freeswitch-1.0.latest.tar.gz on Fedora.
>
>
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From woof at nortel.com  Mon Dec 22 13:38:36 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 22 Dec 2008 16:38:36 -0500
Subject: [Freeswitch-users] Extra loud prompts when transcoded from L16@8000
	to G.722
Message-ID: <op.umk2emi1ssieo1@woof.us.nortel.com>

Woof!

I've noticed that the percieved volume of prompts recorded at L16 at 8000 is much louder (to the point of distortion) when played back via G.722 on Polycom phones, vs when played back via G.711.  The same prompts are also slightly louder when played back on SNOM phones via G.722 vs G.711, but not nearly as obnoxiously loud.  

I'm not sure if FS can do anything about this, it may be a Polycom issue, but is there a way to apply gain (negative in this case!) during the transcoding process so the percieved volume of the prompts is the same no matter which codec is selected?

--Woof!



From brian at freeswitch.org  Mon Dec 22 13:46:14 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 15:46:14 -0600
Subject: [Freeswitch-users] Extra loud prompts when transcoded from
	L16@8000 to G.722
In-Reply-To: <op.umk2emi1ssieo1@woof.us.nortel.com>
References: <op.umk2emi1ssieo1@woof.us.nortel.com>
Message-ID: <3A489B44-DFD2-4D1F-899E-9E7CFEA072D2@freeswitch.org>

When we convert them from 48k we can lower the vol a bit more we are  
already doing it slightly.

/b

On Dec 22, 2008, at 3:38 PM, Andy Spitzer wrote:

> I'm not sure if FS can do anything about this, it may be a Polycom  
> issue, but is there a way to apply gain (negative in this case!)  
> during the transcoding process so the percieved volume of the  
> prompts is the same no matter which codec is selected?




From woof at nortel.com  Mon Dec 22 13:55:05 2008
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 22 Dec 2008 16:55:05 -0500
Subject: [Freeswitch-users] Extra loud prompts when transcoded from
 L16@8000 to G.722
In-Reply-To: <3A489B44-DFD2-4D1F-899E-9E7CFEA072D2@freeswitch.org>
References: <op.umk2emi1ssieo1@woof.us.nortel.com>
	<3A489B44-DFD2-4D1F-899E-9E7CFEA072D2@freeswitch.org>
Message-ID: <op.umk253ebssieo1@woof.us.nortel.com>

Woof!

On Mon, 22 Dec 2008 16:46:14 -0500, Brian West <brian at freeswitch.org> wrote:

> When we convert them from 48k we can lower the vol a bit more we are
> already doing it slightly.
> 

The prompts we are using aren't from the FS set.  It's not a matter of adjusting the prompts, they've work fine for G.711 for years now--it's that when real-time transcoded by FS to G.722 the volume is loud.

Also, consider a call that comes in via G.711 and records a message, saved as L16 at 8000 in a .wav file.  Now play that recording back over G.722.  It's way louder than if played back over G.711.  So depending on which phone you pick up your messages on, the difference in percieved volume is quite dramatic.


--Woof!



From gilbertandrew at me.com  Mon Dec 22 13:56:07 2008
From: gilbertandrew at me.com (Andrew Gilbert)
Date: Mon, 22 Dec 2008 16:56:07 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
 work?
In-Reply-To: <8CB327545F492AF-928-221@WEBMAIL-MA02.sysops.aol.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com>
	<B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com>
	<8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com>
	<8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com>
	<191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com>
	<8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com>
	<191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com>
	<8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com>
	<8CB327545F492AF-928-221@WEBMAIL-MA02.sysops.aol.com>
Message-ID: <37FBF657-259F-4D02-B295-05CEEDA7561F@me.com>

Mark,

This is great, a second box should help, you might want to go back to  
stock setups as well (ie none of the prophecy port changes, etc). This  
will help eliminate configuration error as an issue.

Anthony, will certainly ping you guys when I start testing. Thanks.

Andy



On Dec 22, 2008, at 12:53 PM, mszlazak at aol.com wrote:

> Hi Andy and Anthony.
>
> Thanks Anthony for elaborating more and I'll attempt using another  
> IP on the same box as well.
>
> Also, Prophecy support has asked me first to put one application on  
> a separate box and then get some wireshark data so I'll attempt that  
> also.
>
> Andy, I didn't want to bother you given all those things you had to  
> deal with. Welcome back.
>
> I explored the VMware idea before but was warned that it would not  
> work well with an ASR. This advice came from the Trixbox forums,  
> LumenVox, FreeSwitch and Voxeo.
>
> I understand that what I'm doing goes against the grain (i.e. voip)  
> but frankly my target market really doesn't want anything to do with  
> voip or even internet connectivity from their businesses. Plus there  
> are other issues.
>
> I'll let you know how it goes.
>
> Happy holidays.
>
> Mark.
>
> -----Original Message-----
> From: Andrew Gilbert <gilbertandrew at me.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 22 Dec 2008 7:49 am
> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
> Proxy Media work?
>
> Mark,
>
> Sorry I haven't had much time to help with this either.
>
> But Anthony is offering good advice here. You are either going to  
> have to work out what is going on at SIP/SDP/RTP level through logs  
> and wireshark, or opt fo r a separate ip space. Another option  
> (besides virtual ips) is VMWare or VirtualBox, although VMWare is  
> probably easier to setup and bridge naturally to your host.
>
> Vm's are just so easy anymore and it definitely seems like you are  
> going against the grain right now.
>
> Also - realizing you got here because of the need for ASR. I do have  
> the Lumenvox license, and I was able to compile the module out of  
> SVN. I have not tested anything yet. If things go well I should have  
> some time after the 25th for this. My goal would be to get pizza or  
> something akin to work.
>
> Andy
>
>
> On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:
>
>> I don't really know what your problem is.  I just saw you ask 3  
>> times for help and tried to offer a suggestion.
>> if you start FS with TPORT_LOG=1 you can see all the sip messages  
>> in the console and you could
>> also run wireshark to look at a packet capture.
>>
>> If you use the same IP for media on the same box for 3 programs at  
>> once you may end up with 2 applictions choosing the same media port  
>> etc.
>>
>> It's just a good practice to run every voip program on it's own IP.
>>
>>
>>
>>
>> On Mon, Dec 22, 2008 at 12:44 AM, <mszlazak at aol.com> wrote:
>> Hi Anthony,
>>
>> I actually suggested adding IP's to a Voxeo-Prophecy support person  
>> before but they thought that could be problematic. I went along  
>> with the earlier warning but now you have suggested it again. What  
>> makes everything on the same box tricky?
>>
>> Also, the thing that surprises me a bit is that bypass-media works  
>> but proxy-media or the default doesn't.
>>
>> Would you be kind enough to elaborate.
>>
>> Thanks. Mark.
>>
>>
>>
>> -----Original Message-----
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Sun, 21 Dec 2008 2:49 pm
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
>> Proxy Media work?
>>
>> Try adding more ip to your box and give each thing it's own  
>> dedicated virtual IP.
>> Doing everything on the same box can be tricky.
>>
>>
>> On Sat, Dec 20, 2008 at 2:17 AM, <mszlazak at aol.com> wrote:
>> With the firewall ON or OFF the problem still remains.
>>
>> I've tried 3 different set-ups in a dial plan extension.
>>
>> 1. With  only <action application="set" data="proxy_media=true"/>  
>> before bridging.
>>
>> 2. With only  <action application="set" data="bypass_media=true"/>  
>> before bridging.
>>
>> 3. Neither of the above in the extension.
>>
>> Only 2 with "bypass-media=true" gets the audio across endpoints.
>>
>> Help :-)
>>
>>
>> -----Original Message-----
>> From: mszlazak at aol.com
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Fri, 19 Dec 2008 11:30 am
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
>> Proxy Media work?
>>
>>  With the firewall ON or OFF the problem still remains.
>>
>> I've tried 3 different set-ups in a dial plan extension.
>>
>> 1. With  only <action application="set" data="proxy_media=true"/>  
>> before bridging.
>>
>> 2. With only  <action application="set" data="bypass_media=true"/>  
>> before bridging.
>>
>> 3. Neither of the above in the extension.
>>
>> Only 2 with "proxy-media=true" gets the audio across endpoints.
>>
>> Help :-)
>>
>>
>>
>>
>>
>> 0A
>>
>>
>> -----Original Message-----
>> From: Michael Jerris <mike at jerris.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Fri, 19 Dec 2008 7:49 am
>> Subject: Re: [Freeswitch-users] If Bypass Media works why won't  
>> Proxy Media work?
>>
>> It gives me the impression there is something wrong with your  
>> firewall running on the box.
>>
>> Mike
>>
>> On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
>>
>>> I find it strange that I can have to endpoints get audio went  
>>> using bypass media mode but the audio fails to go across endpoints  
>>> if I use proxy media mode.
>>> I'm trying to pass audio "internally" on the same machine between  
>>> endpoints and have be advis ed that a reason the audio may fail to  
>>> be passed is because there is some RTP timing and IP address/port  
>>> issues.
>>> However, FS has no problem "connecting" ports if i change the mode  
>>> to bypass media. This gives me the impression that something is  
>>> wrong with FS proxy media mode.
>>> Any comments?
>>>
>>> Listen to 350+ music, sports, & news radio stations ? including  
>>> songs for the holidays ? FREE while you browse. Start Listening  
>>> Now!
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch. org
>>
>> =
>> _______________________________________________
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Listen to 350+ music, sports, & news radio stations ? including  
>> songs for the holidays ? FREE while you browse. Start Listening  
>> Now!
>>  _______________________________________________
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>>
>>
>>
>>
>> Freeswitch-users at lists.freeswitch.org
>>
>>
>>
>>
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>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
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>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>>
>>
>>
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>>
>>
>> Listen to 350+ music, sports, & news radio stations ? including  
>> songs for the holidays ? FREE while you browse. Start Listening  
>> Now!
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>  _______________________________________________
>>
>>
>>
>> Freeswitch-users mailing list
>>
>>
>>
>> Freeswitch-users at lists
>>
>> .freeswitch.org
>>
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
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>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> Listen to 350+ music, sports, & news radio stations ? including  
>> songs for the holidays ? FREE while you browse. Start Listening Now!
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> =
> _____________________
> __________________________
>
> Freeswitch-users mailing list
>
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>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> Listen to 350+ music, sports, & news radio stations ? including  
> songs for the holidays ? FREE while you browse. Start Listening Now!
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mszlazak at aol.com  Mon Dec 22 14:35:38 2008
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 22 Dec 2008 17:35:38 -0500
Subject: [Freeswitch-users] If Bypass Media works why won't Proxy Media
	work?
In-Reply-To: <37FBF657-259F-4D02-B295-05CEEDA7561F@me.com>
References: <8CB2FC76AC5AE21-11E8-1CF6@WEBMAIL-MY21.sysops.aol.com><B78FC88A-D3B2-40C7-B5D7-6EEB1B3A924C@jerris.com><8CB30276A477D00-109C-6BA@FWM-D36.sysops.aol.com><8CB3092958BDF1C-914-2073@webmail-db08.sysops.aol.com><191c3a030812211449y468c215fsfe89198fb35feb07@mail.gmail.com><8CB3217F0659C7D-D68-1662@webmail-dx21.sysops.aol.com><191c3a030812220724v1672d3ecn5992a3fe58a04ebc@mail.gmail.com><8C46D0E8-BA86-49D2-89FF-1D825AD9FFBB@me.com><8CB327545F492AF-928-221@WEBMAIL-MA02.sysops.aol.com>
	<37FBF657-259F-4D02-B295-05CEEDA7561F@me.com>
Message-ID: <8CB329CBF686287-C78-317@WEBMAIL-DG02.sim.aol.com>


 Hi Andy,

Yes, it works and I kept my port changes. 

I tried the set-up in both "bypass-media" mode and what I've called "default" mode (i.e. no "bypass-media" nor "proxy-media" settings). Didn't try proxy-mode since the default worked.

The team at Prophecy is almost 100% certain as to what the problem is and which ports are involved. I've sent them the pcap and Prophecy log files.

Also, I was having background "crackling" noise in "bypass-mode" whether I ran FS in the box that it shared with Prophecy or if FS was running from the other box that didn't have Prophecy.

However, when I ran FS from the box without Prophecy in "default" mode then the audio totally cleared up??


 
Mark.


 

-----Original Message-----
From: Andrew Gilbert <gilbertandrew at me.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, 22 Dec 2008 1:56 pm
Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?










Mark,



This is great, a second box should help, you might want to go back to stock setups as well (ie none of the prophecy port changes, etc). This will help eliminate configuration error as an issue.



Anthony, will certainly ping you guys when I start testing. Thanks.




Andy









On Dec 22, 2008, at 12:53 PM, mszlazak at aol.com wrote:



 Hi Andy and Anthony.
 
 Thanks Anthony for elaborating more and I'll attempt using another IP on the same box as well. 
 
 Also, Prophecy support has asked me first to put one
 application on a separate box and then get some wireshark data so I'll attempt that also.
 
 Andy, I didn't want to bother you given all those things you had to deal with. Welcome back.
 
 I explored the VMware idea before but was warned that it would not work well with an ASR. This advice came from the Trixbox forums, LumenVox, FreeSwitch and Voxeo.
 
 I understand that what I'm doing goes against the grain (i.e. voip) but frankly my target market really doesn't want anything to do with voip or even internet connectivity from their businesses. Plus there are other issues.
 
 I'll let you know how it goes.
 
 Happy holidays.
 
 Mark.
 
 
 
 
 -----Original Message-----
 From: Andrew Gilbert <gilbertandrew at me.com>
 To: freeswitch-users at lists.freeswitch.org
 Sent: Mon, 22 Dec 2008 7:49 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 Mark, 

 
 
Sorry I haven't had much time to help with this either.
 

 
 
But Anthony is offering good advice here.?You are either going to have to work out what is going on at SIP/SDP/RTP level through logs and wireshark, or opt fo r a separate ip space. Another option (besides virtual ips) is VMWare or VirtualBox, although VMWare is probably easier to setup and bridge naturally to your host.
 

 
 
Vm's are just so easy anymore and it definitely seems like you are going against the grain right now.
 

 
 
Also - realizing you got here because of the ne
ed for ASR. I do have the Lumenvox license, and I was able to compile the module out of SVN. I have not tested anything yet. If things go well I should have some time after the 25th for this. My goal would be to get pizza or something akin to work.
 

 
 
Andy
 

 
 

 
 
On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote:
 

I don't really know what your problem is.? I just saw you ask 3 times for help and tried to offer a suggestion.
 if you start FS with TPORT_LOG=1 you can see all the sip messages in the console and you could 
 also run wireshark to look at a packet capture.
 
 If you use the same IP for media on the same box for 3 programs at once you may end up with 2 applictions choosing the same media port etc.
 
 It's just a good practice to run every voip program on it's own IP.
 
 
 
 
 
On Mon, Dec 22, 2008 at 12:44 AM,  <mszlazak at aol.com> wrote:
 
 
 Hi Anthony,
 
 I actually suggested adding IP's to a Voxeo-Prophecy support person before but they thought that could be problematic. I went along with the earlier warning but now you have suggested it again. What makes everything on the same box tricky? 
 
 Also, the thing that surprises me a bit is that bypass-media works but proxy-media or the default doesn't. 
 
 Would you be kind enough to elaborate.
 
 Thanks. Mark.
 
 
 
 
 
 
 
 
 
 -----Original Message-----
 From: Anthony Minessale <anthony.minessale
@gmail.com>
 To: freeswitch-users at lists.freeswitch.org
 
 
 

 
 Sent: Sun, 21 Dec 2008 2:49 pm
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 Try adding more ip to your box and give each thing it's own dedicated virtual IP.
 Doing everything on the same box can be tricky.
 
 
 
On Sat, Dec 20, 2008 at 2:17 AM,  <mszlazak at aol.com> wrote:
 
 
With the firewall ON or OFF the problem still remains.
 
 I've tried 3 different set-ups in a dial plan extension.
 
 1. With? only <action application="set" data="proxy_media=true"/> before bridging.
 ??? ??? ??? 
 2. With only? <action application="set" data="bypass_media=true"/> before bridging.
 
 3. Neither of the above in the extension.
 
 
 Only 2 with "bypass-media=true" gets the audio across endpoints.
 
 Help :-)
  
 
 
 
 
 
 
 -----Original Message-----
 From: mszlazak at aol.com
 To: freeswitch-users at lists.freeswitch.org
 
 
 Sent: Fri, 19 Dec 2008 11:30 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 
 
 
?With the firewall ON or OFF the problem still remains.
 
 I've tried 3 different set-ups in a dial plan extension.
 
 1. With? only <action application="set" data="proxy_media=true"/> before bridging.
 ??? ??? ??? 
 2. With only? <action application="set" data="bypass_media=true"/> before bridging.
 
 3.
 Neither of the above in the extension.
 
 Only 2 with "proxy-media=true" gets the audio across endpoints.
 
 Help :-)
 
 
 ? 
 
 
 
 
 0A 
 

 
 
 
 
 
 
 
 -----Original Message-----
 From: Michael Jerris <mike at jerris.com>
 To: freeswitch-users at lists.freeswitch.org
 Sent: Fri, 19 Dec 2008 7:49 am
 Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
 
 
 It gives me the impression there is something wrong with your firewall running on the box. 

 
 
Mike
 

 
 
 
 
On Dec 19, 2008, at 3:03 AM, mszlazak at aol.com wrote:
 
 
I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode.
  I'm trying to pass audio "internally" on the same machine between endpoints and have be advis ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. 
  However, FS has no problem "connecting" ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode.
  Any comments?? 
   

 

Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
  _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch. org
 

 
 
 = 
  
 
_______________________________________________















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Freeswitch-users at lists.freeswitch.org














http://lists.freeswitch.org/mailman/listinfo/freeswitch-users















UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users















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 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 
 

 
   
 
_______________________________________________













Freeswitch-users mailing list







Freeswitch-users at lists.freeswitch.org







http://lists.freeswitch.org/mailman/listinfo/freeswitch-users







UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users







http://www.freeswitch.org








 
   

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 

 
 

 --20
 Anthony Minessale II
 
 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 
 AIM: anthm
 MSN:anthony_minessale at hotmail.com
 GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
 IRC: irc.freenode.net #freeswitch
 
 FreeSWITCH Developer Conference
 sip:888 at conference.freeswitch.org
 iax:guest at conference.freeswitch.org/888
 googletalk:conf+888 at conference.freeswitch.org
 pstn:213-799-1400
 
 
 
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Freeswitch-users mailing list



Freeswitch-users at lists

.freeswitch.org



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



http://www.freeswitch.org




 
 

 

 Listen to 350+ music, sports, & news radio stations ? including songs for the holidays ? FREE while you browse. Start Listening Now! 
 
 
 
 _______________________________________________
 Freeswitch-users mailing list
 Freeswitch-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 



 
 

 -- 
 Anthony Minessale II
 
 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 
 AIM: anthm
 MSN:anthony_minessale at hotmail.com
 GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
 IRC: irc.freenode.net #freeswitch
 
 FreeSWITCH Developer Conference
 sip:888 at conference.freeswitch.org
 iax:guest at conference.freeswitch.org/888
 googletalk:conf+888 at conference.freeswitch.org
 pstn:213-799-1400
 _______________________________________________
 Freeswitch-users mailing list
 Freeswitch
-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 
 
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From jaugenstine at gmail.com  Mon Dec 22 14:51:43 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Mon, 22 Dec 2008 14:51:43 -0800
Subject: [Freeswitch-users] mod_java.so load issue
In-Reply-To: <F009F62E-F171-449D-9B70-4E1090B1DBF0@jerris.com>
References: <207e7a5e0812201104l6280ba16g265486f750f10604@mail.gmail.com>
	<F009F62E-F171-449D-9B70-4E1090B1DBF0@jerris.com>
Message-ID: <207e7a5e0812221451r73decf09j5e6806775290fd5e@mail.gmail.com>

Mike

Could the problem be related to the gcc/g++ version.  The one installed is:
g++ (GCC) 4.3.0 20080428 (Red Hat 4.3.0-8)
I have been trying to look at differences between this server and others
servers I have installed on.  So far, that is the only significant
difference I can identify.  Previously, the gcc version I have built with is
4.0.1.

Jonathan

On Sat, Dec 20, 2008 at 11:33 AM, Michael Jerris <mike at jerris.com> wrote:

> I would suggest cleaning and rebuilding the module.  If that doesn't
> work could we arrange access to the box so I can take a look?
>
> Mike
>
> On Dec 20, 2008, at 2:04 PM, "jonathan augenstine" <jaugenstine at gmail.com
>  > wrote:
>
> > I am installing Freeswitch on Fedora.  I was building/installing the
> > mod_java.so module and I encountered the following load issue:
> >
> > 2008-12-20 10:34:58 [CRIT] switch_loadable_module.c:839
> > switch_loadable_module_load_file() Error Loading module /usr/local/
> > freeswitch/mod/mod_java.so
> > **/usr/local/freeswitch/mod/mod_java.so: invalid ELF header**
> >
> > Is this a build issue?  I am assuming maybe there is a g++ option
> > that is set incorrectly but my searches on Google and looking at gcc
> > docs have not provided a solution.
> >
> > Jonathan
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Mon Dec 22 15:01:02 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 23 Dec 2008 10:01:02 +1100
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <38E75A1D-7391-46A8-BD9C-1C851A019625@freeswitch.org>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<38E75A1D-7391-46A8-BD9C-1C851A019625@freeswitch.org>
Message-ID: <20081222230102.GA7220@jdc.jasonjgw.net>

On Mon, Dec 15, 2008 at 08:45:49AM -0600, Brian West wrote:
> Are you using SVN trunk?  This has been fixed already as far as I  
> remember!!

Just to close out this issue for now, it is possible that the FreeSWITCH
version I was running when this problem occurred included object files from an
earlier build; I ran "make clean" before compiling it, but not "debuild
clean", which cleans all of the libraries as well.

It is also clear that a Hurricane Electric router on the path to New York has
been experiencing issues, which account for some of my IPv6 problems, as the
destination of the calls was connected via tunnel to the New York point of
presence.




From msc at freeswitch.org  Mon Dec 22 15:09:43 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 15:09:43 -0800
Subject: [Freeswitch-users] making outbound call with IPv6
In-Reply-To: <20081222230102.GA7220@jdc.jasonjgw.net>
References: <20081215044113.GA9555@jdc.jasonjgw.net>
	<20081215052518.GA9967@jdc.jasonjgw.net>
	<20081215080105.GA11352@jdc.jasonjgw.net>
	<38E75A1D-7391-46A8-BD9C-1C851A019625@freeswitch.org>
	<20081222230102.GA7220@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812221509k1412a552i37c1a2c040603df@mail.gmail.com>

Thanks for keeping us updated! It's a lot nicer than, "I wonder whatever
happened to Jason and his IPv6 issue?"
:)

-MC

On Mon, Dec 22, 2008 at 3:01 PM, Jason White <jason at jasonjgw.net> wrote:

> On Mon, Dec 15, 2008 at 08:45:49AM -0600, Brian West wrote:
> > Are you using SVN trunk?  This has been fixed already as far as I
> > remember!!
>
> Just to close out this issue for now, it is possible that the FreeSWITCH
> version I was running when this problem occurred included object files from
> an
> earlier build; I ran "make clean" before compiling it, but not "debuild
> clean", which cleans all of the libraries as well.
>
> It is also clear that a Hurricane Electric router on the path to New York
> has
> been experiencing issues, which account for some of my IPv6 problems, as
> the
> destination of the calls was connected via tunnel to the New York point of
> presence.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Dec 22 15:22:30 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Dec 2008 15:22:30 -0800
Subject: [Freeswitch-users] Freeswitch/Sofia configuration problem
In-Reply-To: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>
References: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>
Message-ID: <87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>

Are you using the default config? If you've made any changes at all we'd
need to know about them. Also, can you turn on SIP trace so that we can see
exactly what is coming and going?
Start FS like this:
TPORT_LOG=1 ./freeswitch
Press F8 to put the console in debug mode
then capture the output while you observe the bad behavior
Please put all that, plus any config changes, into a pastebin:
pastebin.freeswitch.org

I'm sure there are people around here who can help you figure out what is
going on.
-MC

On Mon, Dec 22, 2008 at 8:20 AM, Laurent Fabre
<Laurent.Fabre at kirranet.com>wrote:

>
> Hello,
>
> I've been trying to figure out for a few days why my freeswitch instance
> suddenly become insensitive to SIP packets without any warning.
>
> What usually happen is the following :
>
> 1) start just fine in foreground mode and no errors
> 2) wait anywhere between 2 seconds and 20 minutes
> 3) Sofia suddenly decide to reload everything for some reason
> 4) Sofia start processing SIP packets
> 5) work for an hour or so
> 6) Sofia suddenly decide to reload everything for some reason
> 7) become unresponsive again
> 8) goto 2
>
> Both interfaces have public IP addresses assigned in a static manner (no
> DHCP).
>
> I can see the SIP UDP & TCP requests comming from the phones on several
> sites on the wire.
> The SIP TCP requests get RST in reply which is mean :(
>
> There was a point in my setup where it would not happen but since I'm new
> to freeswitch I'm having an hard time backtracking.
>
> I was wondering if iproute/tc and iptables were the culprits but I flushed
> everything (even rebooted without loading the rules) and it still doesn't
> work.
>
> I thought some database was corrupt so I shutdown'd freeswitch and delete
> his db folder, no effect.
>
> My server runs Debian 4.0etch for amd64, built freeswitch from SVN trunk.
>
> Any pointers, help, cure against headaches would be great :)
>
> Regards,
>
> Laurent
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Dec 22 16:01:07 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 18:01:07 -0600
Subject: [Freeswitch-users] Freeswitch/Sofia configuration problem
In-Reply-To: <87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>
References: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>
	<87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>
Message-ID: <D8AB2778-AEF3-48F5-9B95-DC057AA3C581@freeswitch.org>

The problem is FS thinks your IP has changed... you need <param  
name="auto-restart" value="false"/> in sofia.conf.xml

/b

On Dec 22, 2008, at 5:22 PM, Michael Collins wrote:

> Are you using the default config? If you've made any changes at all  
> we'd need to know about them. Also, can you turn on SIP trace so  
> that we can see exactly what is coming and going?
> Start FS like this:
> TPORT_LOG=1 ./freeswitch
> Press F8 to put the console in debug mode
> then capture the output while you observe the bad behavior
> Please put all that, plus any config changes, into a pastebin:  
> pastebin.freeswitch.org
>
> I'm sure there are people around here who can help you figure out  
> what is going on.
> -MC

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From brian at freeswitch.org  Mon Dec 22 16:51:26 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 18:51:26 -0600
Subject: [Freeswitch-users] Extra loud prompts when transcoded from
	L16@8000 to G.722
In-Reply-To: <op.umk253ebssieo1@woof.us.nortel.com>
References: <op.umk2emi1ssieo1@woof.us.nortel.com>
	<3A489B44-DFD2-4D1F-899E-9E7CFEA072D2@freeswitch.org>
	<op.umk253ebssieo1@woof.us.nortel.com>
Message-ID: <7E0C2F90-1AA9-4A15-BAEB-C4A14DA9A932@freeswitch.org>

can you provide me sound file samples?

/b

On Dec 22, 2008, at 3:55 PM, Andy Spitzer wrote:

> Woof!
>
> On Mon, 22 Dec 2008 16:46:14 -0500, Brian West  
> <brian at freeswitch.org> wrote:
>
>> When we convert them from 48k we can lower the vol a bit more we are
>> already doing it slightly.
>>
>
> The prompts we are using aren't from the FS set.  It's not a matter  
> of adjusting the prompts, they've work fine for G.711 for years now-- 
> it's that when real-time transcoded by FS to G.722 the volume is loud.
>
> Also, consider a call that comes in via G.711 and records a message,  
> saved as L16 at 8000 in a .wav file.  Now play that recording back over  
> G.722.  It's way louder than if played back over G.711.  So  
> depending on which phone you pick up your messages on, the  
> difference in percieved volume is quite dramatic.
>
>
> --Woof!

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From john at feith.com  Mon Dec 22 17:10:00 2008
From: john at feith.com (John Wehle)
Date: Mon, 22 Dec 2008 20:10:00 -0500 (EST)
Subject: [Freeswitch-users]  another switch_ivr_set_user() can't find user
Message-ID: <200812230110.mBN1A0t6004432@jwlab.FEITH.COM>

I'm interested in moving some VoIP phones I'm playing with to a different
set of network numbers for various internal reasons.  However, I'm not
looking for multiple FreeSWITCH domains ... I just want one.  Here's what
I did:

  1) On the FreeSWITCH box created a logical network interface and assigned
     it a number from the new set of network numbers.

  2) Changed rtp-ip and sip-ip in internal.xml to use the number assigned
     to the logical network interface.

  3) Per the FAQ set force-register-domain and force-register-db-domain in
     internal.xml to the value for domain in vars.xml (i.e. the main local
     ip v4 address of the FreeSWITCH machine).

  4) Changed the VoIP phone to use a number from the new set.

When I place a call from the VoIP phone FreeSWITCH complains:

  [WARNING] switch_ivr.c:1941 switch_ivr_set_user() can't find user
    [default at 192.168.14.10]
 
where 192.168.14.10 is the number assigned to the logical interface,
however the call goes through / everything seems to work.  I'm able
to place a call to the VoIP phone from openzap without any complaints.

Suggestions / pointers regarding the warning and how to make it happy?

-- John
-------------------------------------------------------------------------
|   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
|    John Wehle    |     Fax: 1-215-540-5495  |                         |
-------------------------------------------------------------------------




From pieter_eduard at biznetnetworks.com  Mon Dec 22 20:00:26 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Tue, 23 Dec 2008 11:00:26 +0700
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>
References: <494F93EA.4080608@biznetnetworks.com>
	<CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>
Message-ID: <4950625A.90100@biznetnetworks.com>

Hi Brian,

I already update the fs to FreeSWITCH Version 1.0.trunk (10906M), and 
the problem still persist .
I already attach the debug log using "console loglevel debug" and post 
it at the jira, and I'm coordinating with the provider regarding 
Anthony's comment about the nat / firewall issue so i hope i could get 
some response from them.

Thank you,

-Pieter-

Brian West wrote:
> I also need you to do this call again with "console loglevel debug" on
> and post it attached to the jira and not inline on the comments please.
>
> /b
> On Dec 22, 2008, at 7:19 AM, Pieter Eduard wrote:
>
>   
>> Hi,
>>
>> I have an enum number, if I call the number from any ip extension ( i
>> use default enum.conf that points to e164.arpa)  then the call goes
>> well
>> to my ATA that registers to my fs box,
>> but if i try to call the number from PLMN, i get the ring at my ATA
>> and
>> if i pick it up, there's no sound.
>>
>> here's my public.xml config :
>>
>> <extension name="enum">
>>        <condition field="destination_number" expression="^12345678">
>>        <action application="set" data="proxy_media=true"/>
>>        <action application="bridge" data="sofia/default/
>> 1000%1.2.3.4"/>
>>        </condition>
>>    </extension>
>>
>> For more detailed debug log, i already submit it on jira :
>> http://jira.freeswitch.org/browse/MODAPP-186
>>
>>
>> regards,
>>
>> -Pieter-
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From jason at jasonjgw.net  Mon Dec 22 20:12:37 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 23 Dec 2008 15:12:37 +1100
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find	user
In-Reply-To: <200812230110.mBN1A0t6004432@jwlab.FEITH.COM>
References: <200812230110.mBN1A0t6004432@jwlab.FEITH.COM>
Message-ID: <20081223041237.GA19348@jdc.jasonjgw.net>

On Mon, Dec 22, 2008 at 08:10:00PM -0500, John Wehle wrote:
> I'm interested in moving some VoIP phones I'm playing with to a different
> set of network numbers for various internal reasons.  However, I'm not
> looking for multiple FreeSWITCH domains ... I just want one.  Here's what
> I did:
> 
>   1) On the FreeSWITCH box created a logical network interface and assigned
>      it a number from the new set of network numbers.
> 
>   2) Changed rtp-ip and sip-ip in internal.xml to use the number assigned
>      to the logical network interface.
> 
>   3) Per the FAQ set force-register-domain and force-register-db-domain in
>      internal.xml to the value for domain in vars.xml (i.e. the main local
>      ip v4 address of the FreeSWITCH machine).
> 
>   4) Changed the VoIP phone to use a number from the new set.
> 

All fine so far.
> When I place a call from the VoIP phone FreeSWITCH complains:
> 
>   [WARNING] switch_ivr.c:1941 switch_ivr_set_user() can't find user
>     [default at 192.168.14.10]
>  
> where 192.168.14.10 is the number assigned to the logical interface,
> however the call goes through / everything seems to work.  

Somewhere in your dial plan, the set_user application is being called with
the above user and domain as parameter.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user

The logs should show you which extensions are being executed in the dial plan
so you can work out how it reached this point and why it's invoking set_user
there. Basically, work through the logic of your dial plan to find out why
this is happening.

I'm sure others will have more specific advice, but, basically, it has to do
with the details of how your dial plan is configured.




From brian at freeswitch.org  Mon Dec 22 20:13:04 2008
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Dec 2008 22:13:04 -0600
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <4950625A.90100@biznetnetworks.com>
References: <494F93EA.4080608@biznetnetworks.com>
	<CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>
	<4950625A.90100@biznetnetworks.com>
Message-ID: <86D1DBA0-07B3-47A7-9CE4-B474166C5E06@freeswitch.org>

First off you opened a jira and assigned it to yourself.. not exactly  
the best way to get it fixed.. secondly what is a PLMN in this  
context?  ... Thirdly... if you notice the bye is sent three times  
which is a big indication that you have a nat problem since its never  
ACK'ed.  Gather up a pcap and attach it to the bug along with a  
console log on "console loglevel debug" attached... please do not  
paste them inline.

/b

On Dec 22, 2008, at 10:00 PM, Pieter Eduard wrote:

> Hi Brian,
>
> I already update the fs to FreeSWITCH Version 1.0.trunk (10906M),  
> and the problem still persist .
> I already attach the debug log using "console loglevel debug" and  
> post it at the jira, and I'm coordinating with the provider  
> regarding Anthony's comment about the nat / firewall issue so i hope  
> i could get some response from them.
>
> Thank you,
>
> -Pieter-




From pieter_eduard at biznetnetworks.com  Mon Dec 22 20:27:37 2008
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Tue, 23 Dec 2008 11:27:37 +0700
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <86D1DBA0-07B3-47A7-9CE4-B474166C5E06@freeswitch.org>
References: <494F93EA.4080608@biznetnetworks.com>	<CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>	<4950625A.90100@biznetnetworks.com>
	<86D1DBA0-07B3-47A7-9CE4-B474166C5E06@freeswitch.org>
Message-ID: <495068B9.5070606@biznetnetworks.com>

Hi Brian,

sorry about the jira, i closed the too much inline log and gonna open a 
new one after i have some results along with the wireshark dump test.

thank you,

-Pieter-

Brian West wrote:
> First off you opened a jira and assigned it to yourself.. not exactly
> the best way to get it fixed.. secondly what is a PLMN in this
> context?  ... Thirdly... if you notice the bye is sent three times
> which is a big indication that you have a nat problem since its never
> ACK'ed.  Gather up a pcap and attach it to the bug along with a
> console log on "console loglevel debug" attached... please do not
> paste them inline.
>
> /b
>
> On Dec 22, 2008, at 10:00 PM, Pieter Eduard wrote:
>
>   
>> Hi Brian,
>>
>> I already update the fs to FreeSWITCH Version 1.0.trunk (10906M),
>> and the problem still persist .
>> I already attach the debug log using "console loglevel debug" and
>> post it at the jira, and I'm coordinating with the provider
>> regarding Anthony's comment about the nat / firewall issue so i hope
>> i could get some response from them.
>>
>> Thank you,
>>
>> -Pieter-
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From carole.olivier at enst.fr  Mon Dec 22 23:03:29 2008
From: carole.olivier at enst.fr (Carole O.)
Date: Mon, 22 Dec 2008 23:03:29 -0800 (PST)
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
References: <21127913.post@talk.nabble.com>
	<2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
Message-ID: <21140461.post@talk.nabble.com>


Hello,

When I do a "show channels" in the cli the channels to the speakers are
listed even if the speakers have stopped transmitting. If I call the
speakers again freeswitch create new channels. If I do a "show channels"
again I can see the old and new ones. If I can keep doing this, each time
new channels are created while the old ones are still there.
I have noticed after 5 minutes the channels that are not used anymore close.
I believed there was a kind of timeout to detect the channels that are not
in use.

What I would like to know is if there is a way to close from these channels
the dialplan .

Thanks,
Carole




Brian West-3 wrote:
> 
> What do you mean they close (hangup) after the 5 minute timeout?
> 
> /b
> 
> On Dec 22, 2008, at 7:35 AM, Carole O. wrote:
> 
>> 1021 and 1022 are the speakers.
>> At the end of the announcement, since there is no noise anymore, the
>> speakers stop listening but they do not send any messages to tell  
>> Freeswitch
>> it can close the opened channels. The channels are closed only after a
>> timeout of 5 minutes.
>>
>> Does anybody know how I could force freeswitch to close all the  
>> channels
>> after the announcement? I have seen there is the application  
>> sched_hangup
>> but when I used it it only closes the channel to the caller and not  
>> the
>> other ones.
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/close-channels-properly-tp21127913p21140461.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From odermann at googlemail.com  Tue Dec 23 00:49:38 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 09:49:38 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
Message-ID: <5e414ed0812230049g16928608md8b7e498cf7b8dce@mail.gmail.com>

hi,

i am quite new to freeswitch and now i finally have fs up and running
as i want with my php scripts to handle the calls.

now that i want to start the service in the near future, i would like
to test the performance of the whole system and the reliability.

what are your experiences and ideas, how i could do it the best way?
and what do i have to look for?

i startet to test with sipp, which seems to be a useful tool, although
i am not sure, how to read the results.

in the switch.conf i set "max-sessions" to 11000 and
"sessions-per-second" to 2000.

in the console i entered: sipp -sn uac xx.xx.xx.xx -s 123456 -r 20 -l
1000 -d 4000 -rtp_echo

the results are as follow:

20.0(4000 ms)/1.000s   5061      51.18 s         1023  xx.xx.xx.xx:5060(UDP)

  20 new calls during 1.001 s period     9 ms scheduler resolution
  807 calls (limit 1000)                 Peak was 807 calls, after 51 s
  0 Running, 807 Paused, 0 Woken up
  0 out-of-call msg (discarded)
  1 open sockets
  75989 Total echo RTP pckts 1st stream  481.979 last period RTP rate (kB/s)
  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)

                                 Messages  Retrans   Timeout   Unexpected-Msg
      INVITE ---------->         1023      0         0
         100 <----------         1023      0                   0
         180 <----------         0         0                   0
         183 <----------         249       0                   0
         200 <----------  E-RTD1 249       0                   0
         ACK ---------->         249       0
       Pause [      0ms]         249                           0
         BYE ---------->         216       0         0
         200 <----------         216       0                   0



  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:072              | 00:08:30:106
  Call Rate              |    0.000 cps              |    4.166 cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |        0
  OutGoing call created  |        0                  |     2125
  Total Call created     |                           |     2125
  Current Call           |        0                  |
-------------------------+---------------------------+--------------------------
  Successful call        |        9                  |     2125
  Failed call            |        0                  |        0
-------------------------+---------------------------+--------------------------
  Response Time 1        | 00:00:00:000              | 00:02:43:140
  Call Length            | 00:04:04:169              | 00:02:47:149



i do not get any errors. but i think that it is strange, that there
are so little calls running at the same time. when i try to call the
same number as sipp does, i have to wait a very long time, till the
call is beeing answered.

why are there so many paused calls waiting for fs to take care of?

is this normal or are there some problems, which i have to take care of?

i want to add, that i am using socket outbound with a number of
php-scripts, which do a lot of call- and dialplan-handling. i am not
using any xml. therefore i am sure, that the results are worse, that
they would be with normal xml dialplans.


thanks for your help and tipps.
dannis



From odermann at googlemail.com  Tue Dec 23 00:52:43 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 09:52:43 +0100
Subject: [Freeswitch-users] Memory question
Message-ID: <5e414ed0812230052n71705a25s6d55e694d609b148@mail.gmail.com>

after doing some testing with fs, i can see in the console, when
entering "top", that fs uses 9.9% of the memory.

when i do some more calls, the used memory will raise - the memory
will not beeing released, till i do a restart of fs.

is this a normal behavior or do i have some problems?


thanks
dennis



From krice at suspicious.org  Tue Dec 23 01:01:44 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:01:44 -0600
Subject: [Freeswitch-users] Memory question
In-Reply-To: <5e414ed0812230052n71705a25s6d55e694d609b148@mail.gmail.com>
Message-ID: <C5760518.4B5C0%krice@suspicious.org>

This is normal behavior... FS allocates memory into pools and re-uses that
same memory over and over... It is quite normal to see memory usage increase
as usage of FS increases to a point where it levels off for that load...  As
the loa decreases memory is not released but used for later when loading
increases again


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 09:52:43 +0100
> To: <Freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Memory question
> 
> after doing some testing with fs, i can see in the console, when
> entering "top", that fs uses 9.9% of the memory.
> 
> when i do some more calls, the used memory will raise - the memory
> will not beeing released, till i do a restart of fs.
> 
> is this a normal behavior or do i have some problems?
> 
> 
> thanks
> dennis
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From krice at suspicious.org  Tue Dec 23 01:03:25 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:03:25 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230049g16928608md8b7e498cf7b8dce@mail.gmail.com>
Message-ID: <C576057D.4B5C2%krice@suspicious.org>

Freeswitch can handle a large volume of call... I suggest you review your
configs to make sure you don't have any of the default or arbitrary other
limits in there... We routinely run > 1500 concurrent calls on dual quad
core hardware at call rates far above what you tested at

Ken


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 09:49:38 +0100
> To: <Freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Performance testing: FS and own App?
> 
> hi,
> 
> i am quite new to freeswitch and now i finally have fs up and running
> as i want with my php scripts to handle the calls.
> 
> now that i want to start the service in the near future, i would like
> to test the performance of the whole system and the reliability.
> 
> what are your experiences and ideas, how i could do it the best way?
> and what do i have to look for?
> 
> i startet to test with sipp, which seems to be a useful tool, although
> i am not sure, how to read the results.
> 
> in the switch.conf i set "max-sessions" to 11000 and
> "sessions-per-second" to 2000.
> 
> in the console i entered: sipp -sn uac xx.xx.xx.xx -s 123456 -r 20 -l
> 1000 -d 4000 -rtp_echo
> 
> the results are as follow:
> 
> 20.0(4000 ms)/1.000s   5061      51.18 s         1023  xx.xx.xx.xx:5060(UDP)
> 
>   20 new calls during 1.001 s period     9 ms scheduler resolution
>   807 calls (limit 1000)                 Peak was 807 calls, after 51 s
>   0 Running, 807 Paused, 0 Woken up
>   0 out-of-call msg (discarded)
>   1 open sockets
>   75989 Total echo RTP pckts 1st stream  481.979 last period RTP rate (kB/s)
>   0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)
> 
>                                  Messages  Retrans   Timeout   Unexpected-Msg
>       INVITE ---------->         1023      0         0
>          100 <----------         1023      0                   0
>          180 <----------         0         0                   0
>          183 <----------         249       0                   0
>          200 <----------  E-RTD1 249       0                   0
>          ACK ---------->         249       0
>        Pause [      0ms]         249                           0
>          BYE ---------->         216       0         0
>          200 <----------         216       0                   0
> 
> 
> 
>   Counter Name           | Periodic value            | Cumulative value
> -------------------------+---------------------------+------------------------
> --
>   Elapsed Time           | 00:00:00:072              | 00:08:30:106
>   Call Rate              |    0.000 cps              |    4.166 cps
> -------------------------+---------------------------+------------------------
> --
>   Incoming call created  |        0                  |        0
>   OutGoing call created  |        0                  |     2125
>   Total Call created     |                           |     2125
>   Current Call           |        0                  |
> -------------------------+---------------------------+------------------------
> --
>   Successful call        |        9                  |     2125
>   Failed call            |        0                  |        0
> -------------------------+---------------------------+------------------------
> --
>   Response Time 1        | 00:00:00:000              | 00:02:43:140
>   Call Length            | 00:04:04:169              | 00:02:47:149
> 
> 
> 
> i do not get any errors. but i think that it is strange, that there
> are so little calls running at the same time. when i try to call the
> same number as sipp does, i have to wait a very long time, till the
> call is beeing answered.
> 
> why are there so many paused calls waiting for fs to take care of?
> 
> is this normal or are there some problems, which i have to take care of?
> 
> i want to add, that i am using socket outbound with a number of
> php-scripts, which do a lot of call- and dialplan-handling. i am not
> using any xml. therefore i am sure, that the results are worse, that
> they would be with normal xml dialplans.
> 
> 
> thanks for your help and tipps.
> dannis
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From odermann at googlemail.com  Tue Dec 23 01:06:04 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 10:06:04 +0100
Subject: [Freeswitch-users] Memory question
In-Reply-To: <C5760518.4B5C0%krice@suspicious.org>
References: <5e414ed0812230052n71705a25s6d55e694d609b148@mail.gmail.com>
	<C5760518.4B5C0%krice@suspicious.org>
Message-ID: <5e414ed0812230106j41acb813xbed5f99727fee92@mail.gmail.com>

thanks for the good explaination and for making me feel better :-)



2008/12/23 Ken Rice <krice at suspicious.org>:
> This is normal behavior... FS allocates memory into pools and re-uses that
> same memory over and over... It is quite normal to see memory usage increase
> as usage of FS increases to a point where it levels off for that load...  As
> the loa decreases memory is not released but used for later when loading
> increases again
>
>
>> From: Dennis <odermann at googlemail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 23 Dec 2008 09:52:43 +0100
>> To: <Freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] Memory question
>>
>> after doing some testing with fs, i can see in the console, when
>> entering "top", that fs uses 9.9% of the memory.
>>
>> when i do some more calls, the used memory will raise - the memory
>> will not beeing released, till i do a restart of fs.
>>
>> is this a normal behavior or do i have some problems?
>>
>>
>> thanks
>> dennis
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From carthick84 at gmail.com  Tue Dec 23 01:10:05 2008
From: carthick84 at gmail.com (B Karthik)
Date: Tue, 23 Dec 2008 14:40:05 +0530
Subject: [Freeswitch-users] Error when building freeswitch on Debian Etch
	64bit.
Message-ID: <c564ccb60812230110j58eb7161rd0c6dc4660d828a5@mail.gmail.com>

Hi,

I am getting the following error when compiling latest Freeswitch with svn
Revision no - 10914 on Debian etch 64bit. Freeswitch version 1.0.1 is
building successfully.

Making all in .
gcc -I/opt/src/freeswitch/src/include
-I/opt/src/freeswitch/libs/libteletone/src -fPIC -Werror -g -ggdb -g -O2
-pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE
-I/opt/src/freeswitch/libs/apr/include
-I/opt/src/freeswitch/libs/apr-util/include -I/opt/src/freeswitch/libs/stfu
-I/opt/src/freeswitch/libs/sqlite -I/opt/src/freeswitch/libs/pcre
-I/opt/src/freeswitch/libs/srtp/include
-I/opt/src/freeswitch/libs/srtp/crypto/include
-I/opt/src/freeswitch/libs/libresample/include -Ilibs/libedit/src
-DSWITCH_HAVE_LIBEDIT -DSWITCH_HAVE_ODBC -Ilibs/libedit/src
-DSWITCH_HAVE_LIBEDIT -g -O2 -Wall -std=c99 -pedantic -o .libs/freeswitch
freeswitch-switch.o  -lm -L/usr/local/lib ./.libs/libfreeswitch.so
libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt -lpthread
libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath
-Wl,/opt/freeswitch/lib
./.libs/libfreeswitch.so: undefined reference to `operator new(unsigned
long)'
./.libs/libfreeswitch.so: undefined reference to `operator delete(void*)'
./.libs/libfreeswitch.so: undefined reference to `__gxx_personality_v0'
./.libs/libfreeswitch.so: undefined reference to `__cxa_pure_virtual'
./.libs/libfreeswitch.so: undefined reference to `vtable for
__cxxabiv1::__class_type_info'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1

Thanks.

B Karthik
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From odermann at googlemail.com  Tue Dec 23 01:31:00 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 10:31:00 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C576057D.4B5C2%krice@suspicious.org>
References: <5e414ed0812230049g16928608md8b7e498cf7b8dce@mail.gmail.com>
	<C576057D.4B5C2%krice@suspicious.org>
Message-ID: <5e414ed0812230131r7c00b1c2ic5b35fdc2d05ceb@mail.gmail.com>

if i do the same test with the 9998, it does not to seem much better:

20.0(4000 ms)/1.000s   5061      68.25 s         1174  xx.xx.xx.xx:5060(UDP)

  0 new calls during 1.008 s period      9 ms scheduler resolution
  710 calls (limit 1000)                 Peak was 782 calls, after 58 s
  0 Running, 710 Paused, 0 Woken up
  0 out-of-call msg (discarded)
  1 open sockets
  595580 Total echo RTP pckts 1st stream 2809.336 last period RTP rate (kB/s)
  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)

                                 Messages  Retrans   Timeout   Unexpected-Msg
      INVITE ---------->         1174      0         0
         100 <----------         1174      0                   0
         180 <----------         0         0                   0
         183 <----------         0         0                   0
         200 <----------  E-RTD1 480       0                   0
         ACK ---------->         480       0
       Pause [      0ms]         480                           0
         BYE ---------->         464       0         0
         200 <----------         464       0                   0

what settings could i review to get more out of the server?

our setup is a new xeon quad core, 4 gb ram and ubuntu 64-bit. we also
entered the ulimit lines and set "manage-presence" to false.


thanks
dennis



2008/12/23 Ken Rice <krice at suspicious.org>:
> Freeswitch can handle a large volume of call... I suggest you review your
> configs to make sure you don't have any of the default or arbitrary other
> limits in there... We routinely run > 1500 concurrent calls on dual quad
> core hardware at call rates far above what you tested at
>
> Ken



From krice at suspicious.org  Tue Dec 23 01:38:02 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:38:02 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230131r7c00b1c2ic5b35fdc2d05ceb@mail.gmail.com>
Message-ID: <C5760D9A.4B5CA%krice@suspicious.org>

There are a number of issues you can be running into... It really depends on
how your app works, what your actual configuration of freeswitch is, disk IO
subsystem, ulimits, etc etc....



> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 10:31:00 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
> 
> if i do the same test with the 9998, it does not to seem much better:
> 
> 20.0(4000 ms)/1.000s   5061      68.25 s         1174  xx.xx.xx.xx:5060(UDP)
> 
>   0 new calls during 1.008 s period      9 ms scheduler resolution
>   710 calls (limit 1000)                 Peak was 782 calls, after 58 s
>   0 Running, 710 Paused, 0 Woken up
>   0 out-of-call msg (discarded)
>   1 open sockets
>   595580 Total echo RTP pckts 1st stream 2809.336 last period RTP rate (kB/s)
>   0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)
> 
>                                  Messages  Retrans   Timeout   Unexpected-Msg
>       INVITE ---------->         1174      0         0
>          100 <----------         1174      0                   0
>          180 <----------         0         0                   0
>          183 <----------         0         0                   0
>          200 <----------  E-RTD1 480       0                   0
>          ACK ---------->         480       0
>        Pause [      0ms]         480                           0
>          BYE ---------->         464       0         0
>          200 <----------         464       0                   0
> 
> what settings could i review to get more out of the server?
> 
> our setup is a new xeon quad core, 4 gb ram and ubuntu 64-bit. we also
> entered the ulimit lines and set "manage-presence" to false.
> 
> 
> thanks
> dennis
> 
> 
> 
> 2008/12/23 Ken Rice <krice at suspicious.org>:
>> Freeswitch can handle a large volume of call... I suggest you review your
>> configs to make sure you don't have any of the default or arbitrary other
>> limits in there... We routinely run > 1500 concurrent calls on dual quad
>> core hardware at call rates far above what you tested at
>> 
>> Ken
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From odermann at googlemail.com  Tue Dec 23 01:43:30 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 10:43:30 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C5760D9A.4B5CA%krice@suspicious.org>
References: <5e414ed0812230131r7c00b1c2ic5b35fdc2d05ceb@mail.gmail.com>
	<C5760D9A.4B5CA%krice@suspicious.org>
Message-ID: <5e414ed0812230143i2f094924s180afea80a66f382@mail.gmail.com>

because the latest result was with the 9998, it can't be out app (at
the moment).

so there are no other typical things or settings i could look for?


2008/12/23 Ken Rice <krice at suspicious.org>:
> There are a number of issues you can be running into... It really depends on
> how your app works, what your actual configuration of freeswitch is, disk IO
> subsystem, ulimits, etc etc....



From krice at suspicious.org  Tue Dec 23 01:48:36 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:48:36 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230143i2f094924s180afea80a66f382@mail.gmail.com>
Message-ID: <C5761014.4B5CE%krice@suspicious.org>

Whats this 9998 to which you refer?


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 10:43:30 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
> 
> because the latest result was with the 9998, it can't be out app (at
> the moment).
> 
> so there are no other typical things or settings i could look for?
> 
> 
> 2008/12/23 Ken Rice <krice at suspicious.org>:
>> There are a number of issues you can be running into... It really depends on
>> how your app works, what your actual configuration of freeswitch is, disk IO
>> subsystem, ulimits, etc etc....
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From odermann at googlemail.com  Tue Dec 23 01:52:09 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 10:52:09 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C5761014.4B5CE%krice@suspicious.org>
References: <5e414ed0812230143i2f094924s180afea80a66f382@mail.gmail.com>
	<C5761014.4B5CE%krice@suspicious.org>
Message-ID: <5e414ed0812230152u78d3ec3ci22961951e8686831@mail.gmail.com>

the 9998 is an extension in the default.xml to test with media flowing
through the line.


2008/12/23 Ken Rice <krice at suspicious.org>:
> Whats this 9998 to which you refer?
>
>
>> From: Dennis <odermann at googlemail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 23 Dec 2008 10:43:30 +0100
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
>>
>> because the latest result was with the 9998, it can't be out app (at
>> the moment).
>>
>> so there are no other typical things or settings i could look for?
>>
>>
>> 2008/12/23 Ken Rice <krice at suspicious.org>:
>>> There are a number of issues you can be running into... It really depends on
>>> how your app works, what your actual configuration of freeswitch is, disk IO
>>> subsystem, ulimits, etc etc....
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From krice at suspicious.org  Tue Dec 23 01:57:40 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 03:57:40 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230152u78d3ec3ci22961951e8686831@mail.gmail.com>
Message-ID: <C5761234.4B5D1%krice@suspicious.org>

Oh! Well who knows how that will affect the performance... I have never
tested it with that... Try the echo tester but be sure you are using the
media refector with sipp or you arent doing anything useful


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 10:52:09 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
> 
> the 9998 is an extension in the default.xml to test with media flowing
> through the line.
> 
> 
> 2008/12/23 Ken Rice <krice at suspicious.org>:
>> Whats this 9998 to which you refer?
>> 
>> 
>>> From: Dennis <odermann at googlemail.com>
>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>> Date: Tue, 23 Dec 2008 10:43:30 +0100
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
>>> 
>>> because the latest result was with the 9998, it can't be out app (at
>>> the moment).
>>> 
>>> so there are no other typical things or settings i could look for?
>>> 
>>> 
>>> 2008/12/23 Ken Rice <krice at suspicious.org>:
>>>> There are a number of issues you can be running into... It really depends
>>>> on
>>>> how your app works, what your actual configuration of freeswitch is, disk
>>>> IO
>>>> subsystem, ulimits, etc etc....
>>> 
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From kawarod at laposte.net  Tue Dec 23 02:02:41 2008
From: kawarod at laposte.net (rod)
Date: Tue, 23 Dec 2008 14:02:41 +0400
Subject: [Freeswitch-users] SIP Headers and use of "" in the header
In-Reply-To: <DE122F9B-6648-4549-96FC-037BFF81BB98@freeswitch.org>
References: <494FB857.9090003@laposte.net>
	<DE122F9B-6648-4549-96FC-037BFF81BB98@freeswitch.org>
Message-ID: <4950B741.8070109@laposte.net>

Thanks guys, it works.

Brian West wrote:
> <action application="export"><![CDATA[sip_h_Diversion=<123456789 at 10.10.10.254 
>  >;reason="unconditional"]]></action>
>
> /b
>
> On Dec 22, 2008, at 9:55 AM, rod wrote:
>
>   
>> Dear All,
>>
>> I've been playing with the freeswitch options for one month now, and
>> I've been able to use it with kamailio working as a registrar.
>>
>> What I'd like to do is to add a diversion header using the following
>> action in the dialplan:
>>    <action application="export"
>> data 
>> ="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="unconditional""/>
>>
>> Please note, that I'd like to put the word "unconditional" between
>> quotes, this is to comply with the SIP gateway to which I'm sending  
>> trafic.
>>
>> But I've been unable to set an escape character to use theses quotes,
>> cause as you may understand, without escape character FS will consider
>> this instruction instead:
>>
>> <action application="export"
>> data="sip_h_Diversion=<123456789 at 10.10.10.254>;reason="
>>
>> cause the quote after reason= is considered as a closing quote for  
>> data=".
>>
>> Is there a way to achieve this.
>>
>> Thanks.
>> rod
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   



From odermann at googlemail.com  Tue Dec 23 02:06:06 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 11:06:06 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C5761234.4B5D1%krice@suspicious.org>
References: <5e414ed0812230152u78d3ec3ci22961951e8686831@mail.gmail.com>
	<C5761234.4B5D1%krice@suspicious.org>
Message-ID: <5e414ed0812230206u68b6d2a1jda3c580a6692d6d4@mail.gmail.com>

sorry, i do not really understand what you mean with: "Try the echo
tester but be sure you are using the media refector with sipp or you
arent doing anything useful".

what is the "echo tester" and what is "media refector" and how could i use it?

i would like to find out, how many people can talk to each other over
the fs server at the same time. a test setup, which simulates real
calls would be very helpful for me. and then i would like to be able
to compare the results with others, to see, if everything is working
as it should.



2008/12/23 Ken Rice <krice at suspicious.org>:
> Oh! Well who knows how that will affect the performance... I have never
> tested it with that... Try the echo tester but be sure you are using the
> media refector with sipp or you arent doing anything useful



From kristjan.ugrin at gmail.com  Tue Dec 23 02:08:07 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Tue, 23 Dec 2008 11:08:07 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
	<191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
Message-ID: <op.uml03t1gatcbto@linux.site>

Today I did some more testing and packet sniffing.
When calling from google talk to google talk, packets are traveling only inside lan, there are some
queries which goes outside, but nothing more.

When using Gtalk to call someone on sip, then those packets are sent outside and I never see them again.
I think this is freeswitch configuration problem (routing?).

Where can I look further to investigate why this happens?

Thanks.


On Mon, 22 Dec 2008 18:30:46 +0100, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> if you see them leave FS and never reach dest.
> It implies a firewall somewhere in between is blocking them.
>
>
> On Mon, Dec 22, 2008 at 10:19 AM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> But what I would like to achieve is something different (quite similar).
>> You type in a message like "call 1001 at 10.99.8.20" and you it would call  
>> a
>> SIP buddy with any local number.
>>
>> In component mode you need to add a buddy everytime for a different sip
>> nr.?
>> Which would mean a lot of numbers if you would like to call more than  
>> one
>> sip nr. in a lan.
>>
>> As for the first issue, there are RTP packets traveling on FS, but never
>> reach destination after they leave our internal network.
>> Do they have to go outside lan even when the call is placed in a lan
>> between gtalk and SIP?
>> Gtalk to gtalk is no problem on same machines...
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>





From krice at suspicious.org  Tue Dec 23 02:09:32 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 23 Dec 2008 04:09:32 -0600
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <5e414ed0812230206u68b6d2a1jda3c580a6692d6d4@mail.gmail.com>
Message-ID: <C57614FC.4B5D5%krice@suspicious.org>

The echo tester was refering to the echo app in freeswitch

The media reflector is part of sipp that just echos media back to the
source... That's the proper way to test media handling capabilities
otherwise you are only seeing 1/2 the media stream


> From: Dennis <odermann at googlemail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Dec 2008 11:06:06 +0100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
> 
> sorry, i do not really understand what you mean with: "Try the echo
> tester but be sure you are using the media refector with sipp or you
> arent doing anything useful".
> 
> what is the "echo tester" and what is "media refector" and how could i use it?
> 
> i would like to find out, how many people can talk to each other over
> the fs server at the same time. a test setup, which simulates real
> calls would be very helpful for me. and then i would like to be able
> to compare the results with others, to see, if everything is working
> as it should.
> 
> 
> 
> 2008/12/23 Ken Rice <krice at suspicious.org>:
>> Oh! Well who knows how that will affect the performance... I have never
>> tested it with that... Try the echo tester but be sure you are using the
>> media refector with sipp or you arent doing anything useful
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From odermann at googlemail.com  Tue Dec 23 02:15:01 2008
From: odermann at googlemail.com (Dennis)
Date: Tue, 23 Dec 2008 11:15:01 +0100
Subject: [Freeswitch-users] Performance testing: FS and own App?
In-Reply-To: <C57614FC.4B5D5%krice@suspicious.org>
References: <5e414ed0812230206u68b6d2a1jda3c580a6692d6d4@mail.gmail.com>
	<C57614FC.4B5D5%krice@suspicious.org>
Message-ID: <5e414ed0812230215i5da30718md3c60c577042a8d@mail.gmail.com>

ah, that sounds interesting. so the echo app is the 9996, right?

how can i start/use the media reflector? is it something, i have to
call sipp with? sorry for this question, but i am very new in this
business.

right now i call sipp with: sipp -sn uac xx.xx.xx.xx -s 123456 -r 50
-l 400 -d 4000 -rtp_echo




2008/12/23 Ken Rice <krice at suspicious.org>:
> The echo tester was refering to the echo app in freeswitch
>
> The media reflector is part of sipp that just echos media back to the
> source... That's the proper way to test media handling capabilities
> otherwise you are only seeing 1/2 the media stream
>
>
>> From: Dennis <odermann at googlemail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 23 Dec 2008 11:06:06 +0100
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] Performance testing: FS and own App?
>>
>> sorry, i do not really understand what you mean with: "Try the echo
>> tester but be sure you are using the media refector with sipp or you
>> arent doing anything useful".
>>
>> what is the "echo tester" and what is "media refector" and how could i use it?
>>
>> i would like to find out, how many people can talk to each other over
>> the fs server at the same time. a test setup, which simulates real
>> calls would be very helpful for me. and then i would like to be able
>> to compare the results with others, to see, if everything is working
>> as it should.
>>
>>
>>
>> 2008/12/23 Ken Rice <krice at suspicious.org>:
>>> Oh! Well who knows how that will affect the performance... I have never
>>> tested it with that... Try the echo tester but be sure you are using the
>>> media refector with sipp or you arent doing anything useful
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From juanbackson at gmail.com  Tue Dec 23 02:39:26 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Tue, 23 Dec 2008 18:39:26 +0800
Subject: [Freeswitch-users] Need help with "No RTP ports available!"
Message-ID: <27c25bc40812230239x7799a6a7l40e41be26a955da7@mail.gmail.com>

Hi,

I am running some stress testings on freeswitch.  When the number of
RTP ports reached around 1248 - 1250, freeswitch starts to pop out "No
RTP ports available!" error:

2008-12-23 13:14:02 [CRIT] sofia_glue.c:562
sofia_glue_tech_choose_port() No RTP ports available!

OS is Centos 5.2 64 bits and freeswitch is compiled with ./configure
--64bit options .  I also followed the wiki to maximize all my ulimit
parameters, but nothing works.  Does anyone know why?  Any help will
be greatly appreciated.

Here are my sys parameters:

[root at localhost bin]# vmstat
procs -----------memory---------- ---swap-- -----io---- --system--
-----cpu------
 r  b   swpd   free   buff  cache   si   so    bi    bo   in   cs us sy id wa st
 0  0      0 2348816 164396 1116604    0    0    14   205 2197  705  3
 3 93  1  0
[root at localhost bin]# free
             total       used       free     shared    buffers     cached
Mem:       3965952    1616748    2349204          0     164396    1116628
-/+ buffers/cache:     335724    3630228
Swap:      2031608          0    2031608
[root at localhost bin]# cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 0
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 1
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 1
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 2
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 2
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 3
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 3
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 4
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 4
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 5
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 5
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 6
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 6
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

processor       : 7
vendor_id       : GenuineIntel
cpu family      : 6
model           : 23
model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
stepping        : 6
cpu MHz         : 1995.054
cache size      : 6144 KB
physical id     : 7
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
bogomips        : 4989.48
clflush size    : 64
cache_alignment : 64
address sizes   : 38 bits physical, 48 bits virtual
power management:

[root at localhost bin]# cat /proc/sys/fs/file-n
cat: /proc/sys/fs/file-n: No such file or directory
[root at localhost bin]# cat /proc/sys/fs/file-
file-max  file-nr
[root at localhost bin]# cat /proc/sys/fs/file-nr
1530    0       372645
[root at localhost bin]# ulimit -a
core file size          (blocks, -c) unlimited
data seg size           (kbytes, -d) unlimited
scheduling priority             (-e) 0
file size               (blocks, -f) unlimited
pending signals                 (-i) unlimited
max locked memory       (kbytes, -l) unlimited
max memory size         (kbytes, -m) unlimited
open files                      (-n) 999999
pipe size            (512 bytes, -p) 8
POSIX message queues     (bytes, -q) unlimited
real-time priority              (-r) 0
stack size              (kbytes, -s) 244
cpu time               (seconds, -t) unlimited
max user processes              (-u) unlimited
virtual memory          (kbytes, -v) unlimited
file locks                      (-x) unlimited
[root at localhost bin]#



From alex at sinapticode.ro  Tue Dec 23 04:25:35 2008
From: alex at sinapticode.ro (Alexandru Nedelcu)
Date: Tue, 23 Dec 2008 14:25:35 +0200
Subject: [Freeswitch-users] Originate retry problem
Message-ID: <1230035135.4982.25.camel@gathern.lan>

Hi,

When I make a unsuccesfull call using session.originate, I'd like to
have a 10 minutes pause and then try again.

For our dialer we are using JS scripts, and setTimeout is not defined,
session.execute("sleep",...) doesn't work because the session has to be
originated first. And I don't really know what originate_retry_sleep_ms
does.

Basically I want a retry as described here, but with a delay between
calls: http://wiki.freeswitch.org/wiki/Busy_Call_Retry





From kristjan.ugrin at gmail.com  Tue Dec 23 06:09:34 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Tue, 23 Dec 2008 15:09:34 +0100
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.uml03t1gatcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
	<191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
	<op.uml03t1gatcbto@linux.site>
Message-ID: <op.ummb98u6atcbto@linux.site>

I've decided to do this properly:
clean fresweetch reinstall.

My worsktation hosts freeswitch + 1 sip phone also running as 1000 (linux - IP 10.99.8.221)
Other windows machine has gtalk with and also a sip phone registered as 1001 (IP 10.99.8.111).

First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works, audio is perfect.
Packets are propery travelling between 10.99.8.221 and 10.99.8.111

Second case :
On windows machine I open gtalk and I open a chat to buddy which is actually a bot logged in on freeswitch (dingaling client mode).
The I started java socket program which listens to icoming messages, after typing into client
"call 1000 at 10.99.8.221" an api command is executed:
"api originate sofia/default/1000 at 10.99.8.221 &bridge(dingaling/gmail.com/gtalk_mail(at)gmail.com)"

A call is placed between gtalk and sip phone 1000, it rings, but when both end answers there is no audio.
After a minute, the call ends itself.
I've attached wireshark dumps from both ends - what is strange is that packets are not trying to got at right IP,
instead they hit some other machine (213.x.x.x), which doesn't make sense.

Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this story):
http://pastebin.com/m75b10388

// I hope the attachments go trough - 17 KB.
test_gtalk_client_side - dump from win machine (gtalk client)
test_sip_client - dump from linux machine (freeswitch and sip phone client)

I hope to get resolved this mistery somehow.

Thank you for all kind answers.
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From ser at man.szczecin.pl  Tue Dec 23 06:20:17 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Tue, 23 Dec 2008 15:20:17 +0100
Subject: [Freeswitch-users] design of XML structure
Message-ID: <1230042017.9989.66.camel@worek.man.szczecin.pl>

Hi,

Wy there is so many ugly constructions in FS configuration files? For
example configuration of a gateway looks like this:

<include>
  <gateway name="asterlink.com">
  <param name="username" value="cluecon"/>
  <param name="realm" value="asterlink.com"/>
  <param name="from-user" value="cluecon"/>
...

This is not good structure design. I know that it gives extreme
flexibility for developers, but config files are for admins not software
developers. IMHO it should look like this:

<include>
  <gateway name="asterlink.com">
  <username>cluecon</username/>
  <realm>asterlink.com</realm/>
  <from-user>cluecon</from-user/>
...

Such structure + schema file would be a great help in configuration
editing (autocompletion and syntax check).

greetings,
Seweryn





From msc at freeswitch.org  Tue Dec 23 06:42:50 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 23 Dec 2008 06:42:50 -0800
Subject: [Freeswitch-users] design of XML structure
In-Reply-To: <1230042017.9989.66.camel@worek.man.szczecin.pl>
References: <1230042017.9989.66.camel@worek.man.szczecin.pl>
Message-ID: <06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>

Hehe, you just stepped on a land mine! There was A LOT of discussion  
about this. The simple fact of the matter is that there was no way to  
make everyone happy so the devs chose a layout that might be "ugly" to  
some. The key is that XML isn't really "pretty" anyway. The point of  
XML is that it needs to be machine readable AND human readable. Since  
humans are (usually) smarter than machines it was decided that machine  
readable is more important. Besides, when all of the GUIs get built  
you won't be hacking XML very much - if at all.

I promise you this: even if you think something is weird or curious  
about how FreeSWITCH works there is ALWAYS a good reason for the  
design decisions. Always. Nothing in FS was left to chance or caprice.

-MC

PS - this was probably a better topic for the -dev list. :)

Sent from my iPhone

On Dec 23, 2008, at 6:20 AM, Seweryn Niemiec <ser at man.szczecin.pl>  
wrote:

> Hi,
>
> Wy there is so many ugly constructions in FS configuration files? For
> example configuration of a gateway looks like this:
>
> <include>
>  <gateway name="asterlink.com">
>  <param name="username" value="cluecon"/>
>  <param name="realm" value="asterlink.com"/>
>  <param name="from-user" value="cluecon"/>
> ...
>
> This is not good structure design. I know that it gives extreme
> flexibility for developers, but config files are for admins not  
> software
> developers. IMHO it should look like this:
>
> <include>
>  <gateway name="asterlink.com">
>  <username>cluecon</username/>
>  <realm>asterlink.com</realm/>
>  <from-user>cluecon</from-user/>
> ...
>
> Such structure + schema file would be a great help in configuration
> editing (autocompletion and syntax check).
>
> greetings,
> Seweryn
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Dec 23 07:03:42 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:03:42 -0600
Subject: [Freeswitch-users] call failed from PLMN to enum number
In-Reply-To: <495068B9.5070606@biznetnetworks.com>
References: <494F93EA.4080608@biznetnetworks.com>	<CE344A98-1ECC-4085-8D42-F3AC83715D33@freeswitch.org>	<4950625A.90100@biznetnetworks.com>
	<86D1DBA0-07B3-47A7-9CE4-B474166C5E06@freeswitch.org>
	<495068B9.5070606@biznetnetworks.com>
Message-ID: <69A3FDE3-1723-47E4-98D6-0153AB36E41E@freeswitch.org>

Na no need to close it... we can just attach everything to the  
existing jira.  ;)

/b

On Dec 22, 2008, at 10:27 PM, Pieter Eduard wrote:

> Hi Brian,
>
> sorry about the jira, i closed the too much inline log and gonna  
> open a new one after i have some results along with the wireshark  
> dump test.
>
> thank you,
>
> -Pieter-




From anthony.minessale at gmail.com  Tue Dec 23 07:03:38 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Dec 2008 09:03:38 -0600
Subject: [Freeswitch-users] Call between gtalk and sip - no audio
In-Reply-To: <op.ummb98u6atcbto@linux.site>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
	<191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
	<op.uml03t1gatcbto@linux.site> <op.ummb98u6atcbto@linux.site>
Message-ID: <191c3a030812230703g3133930djd2cc5ba7853bcdfb@mail.gmail.com>

when 2 devices talk via googles gtalk when they are both behind the same lan
you
are going to have problems.


on thing you can do is make an acl to ignore any candidates that are not
local
add this to your dingaling profile
<param name="candidate-acl" value="myacl"/>

then add myacl to acl.conf.xml that only allows your lan ip.

Turn off all the stun and ext-rtp-ip setting.

OR

use the windows machine from a box that is not on the sam lan behind the
same nat.




On Tue, Dec 23, 2008 at 8:09 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> I've decided to do this properly:
> clean fresweetch reinstall.
>
> My worsktation hosts freeswitch + 1 sip phone also running as 1000 (linux -
> IP 10.99.8.221)
> Other windows machine has gtalk with and also a sip phone registered as
> 1001 (IP 10.99.8.111).
>
> First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works,
> audio is perfect.
> Packets are propery travelling between 10.99.8.221 and 10.99.8.111
>
> Second case :
> On windows machine I open gtalk and I open a chat to buddy which is
> actually a bot logged in on freeswitch (dingaling client mode).
> The I started java socket program which listens to icoming messages, after
> typing into client
> "call 1000 at 10.99.8.221" an api command is executed:
> "api originate sofia/default/1000 at 10.99.8.221 &bridge(dingaling/
> gmail.com/gtalk_mail(at)gmail.com<http://gmail.com/gtalk_mail%28at%29gmail.com>
> )"
>
> A call is placed between gtalk and sip phone 1000, it rings, but when both
> end answers there is no audio.
> After a minute, the call ends itself.
> I've attached wireshark dumps from both ends - what is strange is that
> packets are not trying to got at right IP,
> instead they hit some other machine (213.x.x.x), which doesn't make sense.
>
> Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this
> story):
> http://pastebin.com/m75b10388
>
> // I hope the attachments go trough - 17 KB.
> test_gtalk_client_side - dump from win machine (gtalk client)
> test_sip_client - dump from linux machine (freeswitch and sip phone client)
>
> I hope to get resolved this mistery somehow.
>
> Thank you for all kind answers.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Tue Dec 23 07:04:53 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:04:53 -0600
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <21140461.post@talk.nabble.com>
References: <21127913.post@talk.nabble.com>
	<2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
	<21140461.post@talk.nabble.com>
Message-ID: <490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org>

Well in this context the phones need to hangup... they aren't going to  
do so automatically.  So you'll need to hang up on them or they will  
need to hangup... or you can kick everyone from the conference with an  
api command.

/b

On Dec 23, 2008, at 1:03 AM, Carole O. wrote:

>
> Hello,
>
> When I do a "show channels" in the cli the channels to the speakers  
> are
> listed even if the speakers have stopped transmitting. If I call the
> speakers again freeswitch create new channels. If I do a "show  
> channels"
> again I can see the old and new ones. If I can keep doing this, each  
> time
> new channels are created while the old ones are still there.
> I have noticed after 5 minutes the channels that are not used  
> anymore close.
> I believed there was a kind of timeout to detect the channels that  
> are not
> in use.
>
> What I would like to know is if there is a way to close from these  
> channels
> the dialplan .
>
> Thanks,
> Carole




From brian at freeswitch.org  Tue Dec 23 07:06:30 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:06:30 -0600
Subject: [Freeswitch-users] Error when building freeswitch on Debian
	Etch 64bit.
In-Reply-To: <c564ccb60812230110j58eb7161rd0c6dc4660d828a5@mail.gmail.com>
References: <c564ccb60812230110j58eb7161rd0c6dc4660d828a5@mail.gmail.com>
Message-ID: <7B08D097-C40B-419F-BB29-C8E192A8FB87@freeswitch.org>

Please install c++ compiler.

/b

On Dec 23, 2008, at 3:10 AM, B Karthik wrote:

> Hi,
>
> I am getting the following error when compiling latest Freeswitch  
> with svn Revision no - 10914 on Debian etch 64bit. Freeswitch  
> version 1.0.1 is building successfully.
>
> Making all in .
> gcc -I/opt/src/freeswitch/src/include -I/opt/src/freeswitch/libs/ 
> libteletone/src -fPIC -Werror -g -ggdb -g -O2 -pthread -DLINUX=2 - 
> D_REENTRANT -D_GNU_SOURCE -I/opt/src/freeswitch/libs/apr/include -I/ 
> opt/src/freeswitch/libs/apr-util/include -I/opt/src/freeswitch/libs/ 
> stfu -I/opt/src/freeswitch/libs/sqlite -I/opt/src/freeswitch/libs/ 
> pcre -I/opt/src/freeswitch/libs/srtp/include -I/opt/src/freeswitch/ 
> libs/srtp/crypto/include -I/opt/src/freeswitch/libs/libresample/ 
> include -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -DSWITCH_HAVE_ODBC - 
> Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -Wall -std=c99 - 
> pedantic -o .libs/freeswitch freeswitch-switch.o  -lm -L/usr/local/ 
> lib ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl - 
> lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,-- 
> rpath -Wl,/opt/freeswitch/lib
> ./.libs/libfreeswitch.so: undefined reference to `operator  
> new(unsigned long)'
> ./.libs/libfreeswitch.so: undefined reference to `operator  
> delete(void*)'
> ./.libs/libfreeswitch.so: undefined reference to  
> `__gxx_personality_v0'
> ./.libs/libfreeswitch.so: undefined reference to `__cxa_pure_virtual'
> ./.libs/libfreeswitch.so: undefined reference to `vtable for  
> __cxxabiv1::__class_type_info'
> collect2: ld returned 1 exit status
> make[2]: *** [freeswitch] Error 1
>
> Thanks.
>
> B Karthik
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Tue Dec 23 07:07:25 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:07:25 -0600
Subject: [Freeswitch-users] Need help with "No RTP ports available!"
In-Reply-To: <27c25bc40812230239x7799a6a7l40e41be26a955da7@mail.gmail.com>
References: <27c25bc40812230239x7799a6a7l40e41be26a955da7@mail.gmail.com>
Message-ID: <5D155529-6D4F-4AB9-9A35-45980DF4A34F@freeswitch.org>

How exactly are you load testing this?  Can you provide us an example?
/b

On Dec 23, 2008, at 4:39 AM, Juan Backson wrote:

> Hi,
>
> I am running some stress testings on freeswitch.  When the number of
> RTP ports reached around 1248 - 1250, freeswitch starts to pop out "No
> RTP ports available!" error:
>
> 2008-12-23 13:14:02 [CRIT] sofia_glue.c:562
> sofia_glue_tech_choose_port() No RTP ports available!
>
> OS is Centos 5.2 64 bits and freeswitch is compiled with ./configure
> --64bit options .  I also followed the wiki to maximize all my ulimit
> parameters, but nothing works.  Does anyone know why?  Any help will
> be greatly appreciated.
>
> Here are my sys parameters:
>
> [root at localhost bin]# vmstat
> procs -----------memory---------- ---swap-- -----io---- --system--
> -----cpu------
> r  b   swpd   free   buff  cache   si   so    bi    bo   in   cs us  
> sy id wa st
> 0  0      0 2348816 164396 1116604    0    0    14   205 2197  705  3
> 3 93  1  0
> [root at localhost bin]# free
>             total       used       free     shared    buffers      
> cached
> Mem:       3965952    1616748    2349204          0     164396     
> 1116628
> -/+ buffers/cache:     335724    3630228
> Swap:      2031608          0    2031608
> [root at localhost bin]# cat /proc/cpuinfo
> processor       : 0
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 0
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 1
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 1
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 2
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 2
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 3
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 3
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 4
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 4
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 5
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 5
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 6
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 6
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> processor       : 7
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 23
> model name      : Intel(R) Xeon(R) CPU           E5405  @ 2.00GHz
> stepping        : 6
> cpu MHz         : 1995.054
> cache size      : 6144 KB
> physical id     : 7
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc
> pni monitor ds_cpl vmx tm2 cx16 xtpr lahf_lm
> bogomips        : 4989.48
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 38 bits physical, 48 bits virtual
> power management:
>
> [root at localhost bin]# cat /proc/sys/fs/file-n
> cat: /proc/sys/fs/file-n: No such file or directory
> [root at localhost bin]# cat /proc/sys/fs/file-
> file-max  file-nr
> [root at localhost bin]# cat /proc/sys/fs/file-nr
> 1530    0       372645
> [root at localhost bin]# ulimit -a
> core file size          (blocks, -c) unlimited
> data seg size           (kbytes, -d) unlimited
> scheduling priority             (-e) 0
> file size               (blocks, -f) unlimited
> pending signals                 (-i) unlimited
> max locked memory       (kbytes, -l) unlimited
> max memory size         (kbytes, -m) unlimited
> open files                      (-n) 999999
> pipe size            (512 bytes, -p) 8
> POSIX message queues     (bytes, -q) unlimited
> real-time priority              (-r) 0
> stack size              (kbytes, -s) 244
> cpu time               (seconds, -t) unlimited
> max user processes              (-u) unlimited
> virtual memory          (kbytes, -v) unlimited
> file locks                      (-x) unlimited
> [root at localhost bin]#
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Tue Dec 23 07:13:42 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:13:42 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find	user
In-Reply-To: <20081223041237.GA19348@jdc.jasonjgw.net>
References: <200812230110.mBN1A0t6004432@jwlab.FEITH.COM>
	<20081223041237.GA19348@jdc.jasonjgw.net>
Message-ID: <55A22F7C-BEF9-4238-93CE-C5881C02B7DA@freeswitch.org>

You don't have a default user in domain 192.168.14.10, in the default  
config I used this so that you can set some vars on every call with  
one call to set_user and it would set all the vars from the default  
user on the current session.  Its best to rip it out the set_user call  
if you have modified things.

/b

On Dec 22, 2008, at 10:12 PM, Jason White wrote:

> All fine so far.
>> When I place a call from the VoIP phone FreeSWITCH complains:
>>
>>  [WARNING] switch_ivr.c:1941 switch_ivr_set_user() can't find user
>>    [default at 192.168.14.10]
>>
>> where 192.168.14.10 is the number assigned to the logical interface,
>> however the call goes through / everything seems to work.
>
> Somewhere in your dial plan, the set_user application is being  
> called with
> the above user and domain as parameter.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
>
> The logs should show you which extensions are being executed in the  
> dial plan
> so you can work out how it reached this point and why it's invoking  
> set_user
> there. Basically, work through the logic of your dial plan to find  
> out why
> this is happening.
>
> I'm sure others will have more specific advice, but, basically, it  
> has to do
> with the details of how your dial plan is configured.

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From kristjan.ugrin at gmail.com  Tue Dec 23 07:15:47 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Tue, 23 Dec 2008 16:15:47 +0100
Subject: [Freeswitch-users] [SOLVED] Call between gtalk and sip - no
	audio
In-Reply-To: <191c3a030812230703g3133930djd2cc5ba7853bcdfb@mail.gmail.com>
References: <op.umki4gzjatcbto@linux.site>
	<191c3a030812220719i67f40228m467f8abc5c71b4ff@mail.gmail.com>
	<op.umklwrrxatcbto@linux.site>
	<191c3a030812220802u63974b57t432d12a06bba5066@mail.gmail.com>
	<op.umknnbetatcbto@linux.site>
	<191c3a030812220930s52c9ea05o8adfa74e3e059a63@mail.gmail.com>
	<op.uml03t1gatcbto@linux.site> <op.ummb98u6atcbto@linux.site>
	<191c3a030812230703g3133930djd2cc5ba7853bcdfb@mail.gmail.com>
Message-ID: <op.ummfclkqatcbto@linux.site>

Thanks, commenting ext-rtp fixed my issue.
In case of further problems I'll do what you suggested.

Thank you again for all help.

On Tue, 23 Dec 2008 16:03:38 +0100, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> when 2 devices talk via googles gtalk when they are both behind the same  
> lan
> you
> are going to have problems.
>
>
> on thing you can do is make an acl to ignore any candidates that are not
> local
> add this to your dingaling profile
> <param name="candidate-acl" value="myacl"/>
>
> then add myacl to acl.conf.xml that only allows your lan ip.
>
> Turn off all the stun and ext-rtp-ip setting.
>
> OR
>
> use the windows machine from a box that is not on the sam lan behind the
> same nat.
>
>
>
>
> On Tue, Dec 23, 2008 at 8:09 AM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> I've decided to do this properly:
>> clean fresweetch reinstall.
>>
>> My worsktation hosts freeswitch + 1 sip phone also running as 1000  
>> (linux -
>> IP 10.99.8.221)
>> Other windows machine has gtalk with and also a sip phone registered as
>> 1001 (IP 10.99.8.111).
>>
>> First case - SIP to SIP. Calling from 1000 to 1001 and vice versa works,
>> audio is perfect.
>> Packets are propery travelling between 10.99.8.221 and 10.99.8.111
>>
>> Second case :
>> On windows machine I open gtalk and I open a chat to buddy which is
>> actually a bot logged in on freeswitch (dingaling client mode).
>> The I started java socket program which listens to icoming messages,  
>> after
>> typing into client
>> "call 1000 at 10.99.8.221" an api command is executed:
>> "api originate sofia/default/1000 at 10.99.8.221 &bridge(dingaling/
>> gmail.com/gtalk_mail(at)gmail.com<http://gmail.com/gtalk_mail%28at%29gmail.com>
>> )"
>>
>> A call is placed between gtalk and sip phone 1000, it rings, but when  
>> both
>> end answers there is no audio.
>> After a minute, the call ends itself.
>> I've attached wireshark dumps from both ends - what is strange is that
>> packets are not trying to got at right IP,
>> instead they hit some other machine (213.x.x.x), which doesn't make  
>> sense.
>>
>> Fresh log from freeswitch (I don't know why 213.x.x.x gets mixed in this
>> story):
>> http://pastebin.com/m75b10388
>>
>> // I hope the attachments go trough - 17 KB.
>> test_gtalk_client_side - dump from win machine (gtalk client)
>> test_sip_client - dump from linux machine (freeswitch and sip phone  
>> client)
>>
>> I hope to get resolved this mistery somehow.
>>
>> Thank you for all kind answers.
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>



-- 
Porn - the reason you need a new hard drive.





From ser at man.szczecin.pl  Tue Dec 23 07:19:36 2008
From: ser at man.szczecin.pl (Seweryn Niemiec)
Date: Tue, 23 Dec 2008 16:19:36 +0100
Subject: [Freeswitch-users] design of XML structure
In-Reply-To: <06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
References: <1230042017.9989.66.camel@worek.man.szczecin.pl>
	<06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
Message-ID: <1230045576.9989.78.camel@worek.man.szczecin.pl>

On Tue, 2008-12-23 at 06:42 -0800, Michael S Collins wrote:
> Hehe, you just stepped on a land mine! There was A LOT of discussion  

ok. I promise that this my last post on this subject :)

> about this. The simple fact of the matter is that there was no way to  
> make everyone happy so the devs chose a layout that might be "ugly" to  
> some. The key is that XML isn't really "pretty" anyway. The point of  
> XML is that it needs to be machine readable AND human readable. Since  
> humans are (usually) smarter than machines it was decided that machine  
> readable is more important. 

I like XML actually. IMHO xml with good editor and schema... there is 
nothing better for config writer.

-- 
greetings,
Seweryn




From msc at freeswitch.org  Tue Dec 23 07:52:16 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 23 Dec 2008 07:52:16 -0800
Subject: [Freeswitch-users] Originate retry problem
In-Reply-To: <1230035135.4982.25.camel@gathern.lan>
References: <1230035135.4982.25.camel@gathern.lan>
Message-ID: <AF39A731-D5D0-4543-8B95-BB271A660136@freeswitch.org>

Have you checked out 'sched_api'?

-MC

Sent from my iPhone

On Dec 23, 2008, at 4:25 AM, Alexandru Nedelcu <alex at sinapticode.ro>  
wrote:

> Hi,
>
> When I make a unsuccesfull call using session.originate, I'd like to
> have a 10 minutes pause and then try again.
>
> For our dialer we are using JS scripts, and setTimeout is not defined,
> session.execute("sleep",...) doesn't work because the session has to  
> be
> originated first. And I don't really know what  
> originate_retry_sleep_ms
> does.
>
> Basically I want a retry as described here, but with a delay between
> calls: http://wiki.freeswitch.org/wiki/Busy_Call_Retry
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Tue Dec 23 07:53:39 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 23 Dec 2008 07:53:39 -0800
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org>
References: <21127913.post@talk.nabble.com>
	<2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
	<21140461.post@talk.nabble.com>
	<490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org>
Message-ID: <97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org>

Carole,

Are you calling the hangup app from the Dialplan?

-MC

Sent from my iPhone

On Dec 23, 2008, at 7:04 AM, Brian West <brian at freeswitch.org> wrote:

> Well in this context the phones need to hangup... they aren't going to
> do so automatically.  So you'll need to hang up on them or they will
> need to hangup... or you can kick everyone from the conference with an
> api command.
>
> /b
>
> On Dec 23, 2008, at 1:03 AM, Carole O. wrote:
>
>>
>> Hello,
>>
>> When I do a "show channels" in the cli the channels to the speakers
>> are
>> listed even if the speakers have stopped transmitting. If I call the
>> speakers again freeswitch create new channels. If I do a "show
>> channels"
>> again I can see the old and new ones. If I can keep doing this, each
>> time
>> new channels are created while the old ones are still there.
>> I have noticed after 5 minutes the channels that are not used
>> anymore close.
>> I believed there was a kind of timeout to detect the channels that
>> are not
>> in use.
>>
>> What I would like to know is if there is a way to close from these
>> channels
>> the dialplan .
>>
>> Thanks,
>> Carole
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Dec 23 07:57:07 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 09:57:07 -0600
Subject: [Freeswitch-users] close channels properly
In-Reply-To: <97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org>
References: <21127913.post@talk.nabble.com>
	<2E97F65B-6903-4C35-99C7-76799D9B72AD@freeswitch.org>
	<21140461.post@talk.nabble.com>
	<490132B3-A767-4441-BB98-A57F3CBFE51D@freeswitch.org>
	<97D97C56-AECF-4F43-8432-8DE12A652D0C@freeswitch.org>
Message-ID: <E9922E74-A4F4-47CE-B841-A1B31B526564@freeswitch.org>

I think Carole is calling a group of people into a conference..  
leaving and expecting everyone to get kicked.

/b

On Dec 23, 2008, at 9:53 AM, Michael S Collins wrote:

> Carole,
>
> Are you calling the hangup app from the Dialplan?
>
> -MC
>
> Sent from my iPhone




From chris.chen2004 at gmail.com  Tue Dec 23 10:12:16 2008
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Tue, 23 Dec 2008 13:12:16 -0500
Subject: [Freeswitch-users] Latest SVN trunk r10919 failed to build on OS X
Message-ID: <507898380812231012t4369dbcal4cfa6878be2f7c41@mail.gmail.com>

Hi all, if you have the same problem as me failing to build the latest trunk
on OS X 10.5.6
please see the error message below:

making all in .
Compiling src/switch_xml.c ...
cc1: warnings being treated as errors
src/switch_xml.c: In function 'switch_xml_find_child_multi':
src/switch_xml.c:310: warning: 'value' may be used uninitialized in this
functio
n
make[2]: *** [libfreeswitch_la-switch_xml.lo] Error 1
Making all in src
Making all in mod

making all mod_cdr_csv
make[5]: *** No rule to make target
`/Users/yunzhangchen/freeswitch/libfreeswitc
h.la', needed by `mod_cdr_csv.so'.  Stop.
make[4]: *** [all] Error 1
make[3]: *** [mod_cdr_csv-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2


Thanks

Chris
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From brian at freeswitch.org  Tue Dec 23 10:17:57 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 12:17:57 -0600
Subject: [Freeswitch-users] Latest SVN trunk r10919 failed to build on
	OS X
In-Reply-To: <507898380812231012t4369dbcal4cfa6878be2f7c41@mail.gmail.com>
References: <507898380812231012t4369dbcal4cfa6878be2f7c41@mail.gmail.com>
Message-ID: <0E399C73-2FFC-4769-A4E2-05B3F75F7ED3@freeswitch.org>

Update and try again.

/b

On Dec 23, 2008, at 12:12 PM, Chris Chen wrote:

> Hi all, if you have the same problem as me failing to build the  
> latest trunk on OS X 10.5.6
> please see the error message below:
>
> making all in .
> Compiling src/switch_xml.c ...
> cc1: warnings being treated as errors
> src/switch_xml.c: In function 'switch_xml_find_child_multi':
> src/switch_xml.c:310: warning: 'value' may be used uninitialized in  
> this functio
> n
> make[2]: *** [libfreeswitch_la-switch_xml.lo] Error 1
> Making all in src
> Making all in mod
>
> making all mod_cdr_csv
> make[5]: *** No rule to make target `/Users/yunzhangchen/freeswitch/ 
> libfreeswitc
> h.la', needed by `mod_cdr_csv.so'.  Stop.
> make[4]: *** [all] Error 1
> make[3]: *** [mod_cdr_csv-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
>  +-------- FreeSWITCH Build Complete -----------+
>  + FreeSWITCH has been successfully built.      +
>  + Install by running:                          +
>  +                                              +
>  +               make install                   +
>  +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
>
> Thanks
>
> Chris
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Tue Dec 23 14:31:15 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 24 Dec 2008 09:31:15 +1100
Subject: [Freeswitch-users] design of XML structure
In-Reply-To: <06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
References: <1230042017.9989.66.camel@worek.man.szczecin.pl>
	<06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
Message-ID: <20081223223115.GA6083@jdc.jasonjgw.net>

On Tue, Dec 23, 2008 at 06:42:50AM -0800, Michael S Collins wrote:
> Besides, when all of the GUIs get built  
> you won't be hacking XML very much - if at all.

And the XML will still be there for those of us who prefer editing
configuration files to using GUIs. I'm very much a Unix shell type of person.




From john at feith.com  Tue Dec 23 14:51:38 2008
From: john at feith.com (John Wehle)
Date: Tue, 23 Dec 2008 17:51:38 -0500 (EST)
Subject: [Freeswitch-users]  another switch_ivr_set_user() can't find user
Message-ID: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>

> You don't have a default user in domain 192.168.14.10, in the default  
> config I used this so that you can set some vars on every call with

Thanks for pointing it out and explaining the purpose.

It looks like the domain is coming from set_domain in default.xml
which gets it from sip_auth_realm.  I guess the question is if
force-register-domain is being used then:

  a) Should sip_auth_realm be set by FreeSWITCH to the value associated
     with force-register-domain

  b) or should set_domain in default.xml simply check for force-register-domain
     when setting domain?

-- John
-------------------------------------------------------------------------
|   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
|    John Wehle    |     Fax: 1-215-540-5495  |                         |
-------------------------------------------------------------------------




From brian at freeswitch.org  Tue Dec 23 15:22:27 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 17:22:27 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find user
In-Reply-To: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
References: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
Message-ID: <31BD69DD-90C5-4D7D-8826-7EE98B335F42@freeswitch.org>

You have to remember the default assumes a lot.  You go to changing  
things you have to then change the way things are assumed.

/b

On Dec 23, 2008, at 4:51 PM, John Wehle wrote:

>> You don't have a default user in domain 192.168.14.10, in the default
>> config I used this so that you can set some vars on every call with
>
> Thanks for pointing it out and explaining the purpose.
>
> It looks like the domain is coming from set_domain in default.xml
> which gets it from sip_auth_realm.  I guess the question is if
> force-register-domain is being used then:
>
>  a) Should sip_auth_realm be set by FreeSWITCH to the value associated
>     with force-register-domain
>
>  b) or should set_domain in default.xml simply check for force- 
> register-domain
>     when setting domain?
>
> -- John
> -------------------------------------------------------------------------
> |   Feith Systems  |   Voice: 1-215-646-8000  |  Email:  
> john at feith.com  |
> |    John Wehle    |     Fax: 1-215-540-5495   
> |                         |
> -------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jason at jasonjgw.net  Tue Dec 23 15:27:31 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 24 Dec 2008 10:27:31 +1100
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find	user
In-Reply-To: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
References: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
Message-ID: <20081223232731.GA8541@jdc.jasonjgw.net>

On Tue, Dec 23, 2008 at 05:51:38PM -0500, John Wehle wrote:
 
> It looks like the domain is coming from set_domain in default.xml
> which gets it from sip_auth_realm.  I guess the question is if
> force-register-domain is being used then:
> 
>   a) Should sip_auth_realm be set by FreeSWITCH to the value associated
>      with force-register-domain
> 
>   b) or should set_domain in default.xml simply check for force-register-domain

As I understand it, that's a choice you need to make in designing your dial
plan.

I'm currently confronted with the same issue and, being new to FreeSWITCH, I'm
not sure what would be the best solution.

In my case, there is a SIP phone that needs to contact FreeSWITCH by its IPv4
address - there's no A record for the FreeSWITCH box, and the address is
192.168.0.2 on the local LAN. The FreeSWITCH machine does have an AAAA record
in DNS, so any incoming IPv6 call will address it by its fully-qualified
domain name. Not surprisingly, the logic in the supplied external.xml dialplan
configuration results in IPv6 incoming calls to local extensions failing, due
to the wrong domain name, after the transfer to default.xml when the bridge
application is executed. By default, of course, the domain is the IPv4
address, which is fine for IPv4 calls.

There are ways around this, of course, but I haven't worked out what would be
the best to implement.




From brian at freeswitch.org  Tue Dec 23 15:32:13 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Dec 2008 17:32:13 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find	user
In-Reply-To: <20081223232731.GA8541@jdc.jasonjgw.net>
References: <200812232251.mBNMpcwd005395@jwlab.FEITH.COM>
	<20081223232731.GA8541@jdc.jasonjgw.net>
Message-ID: <97B3F1B3-E5C0-4EFB-A32D-38E208D40EBB@freeswitch.org>

The default is just an example that tries to get you started... the  
key thing you need to remember for ALL calls coming in that aren't  
authenticated the domain_name variable needs to be set before you  
transfer into the default context.

/b

On Dec 23, 2008, at 5:27 PM, Jason White wrote:

>
> There are ways around this, of course, but I haven't worked out what  
> would be
> the best to implement.




From msc at freeswitch.org  Tue Dec 23 16:08:48 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Dec 2008 16:08:48 -0800
Subject: [Freeswitch-users] design of XML structure
In-Reply-To: <20081223223115.GA6083@jdc.jasonjgw.net>
References: <1230042017.9989.66.camel@worek.man.szczecin.pl>
	<06B0FC64-90F2-4826-8922-6B29DBE12D36@freeswitch.org>
	<20081223223115.GA6083@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812231608x41455914gc060ec5d246e5375@mail.gmail.com>

yeah, vim and emacs will always be available as your GUI. ;)
-MC

On Tue, Dec 23, 2008 at 2:31 PM, Jason White <jason at jasonjgw.net> wrote:

> On Tue, Dec 23, 2008 at 06:42:50AM -0800, Michael S Collins wrote:
> > Besides, when all of the GUIs get built
> > you won't be hacking XML very much - if at all.
>
> And the XML will still be there for those of us who prefer editing
> configuration files to using GUIs. I'm very much a Unix shell type of
> person.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Dec 23 16:46:44 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Dec 2008 16:46:44 -0800
Subject: [Freeswitch-users] FS Dev Appreciation
Message-ID: <87f2f3b90812231646y33d92da6g1beb3295c197a33c@mail.gmail.com>

FYI,

If anyone would like to show their appreciation for the FS Dev team please
email me off list so we can talk about options. I think we can all agree
that Tony & Co. have worked very hard to make FreeSWITCH a success for all
of us. Let's see if we can give a little love back to the core team!

Thanks,
Michael
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From frank at impactfax.com  Tue Dec 23 18:06:45 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 23 Dec 2008 21:06:45 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
Message-ID: <453101c9656c$45ab0900$33014c0a@ws4>

Can this command be used to run a bash script?   
I wanted to do some sox processing on some recordings after the bridge
ends and thought I should use this command.  But would like to do it in
bash. 
 
Is there a better way?
 
If this is the right way, what is the syntax for calling the bash script
with some arguments?
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From jason at jasonjgw.net  Tue Dec 23 18:25:36 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 24 Dec 2008 13:25:36 +1100
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <453101c9656c$45ab0900$33014c0a@ws4>
References: <453101c9656c$45ab0900$33014c0a@ws4>
Message-ID: <20081224022536.GA6240@jdc.jasonjgw.net>

Frank @ Impact <frank at impactfax.com> wrote:
> Can this command be used to run a bash script?   

Based on information at the wiki, this should be possible; use the system
command.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system




From msc at freeswitch.org  Tue Dec 23 22:00:24 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 23 Dec 2008 22:00:24 -0800
Subject: [Freeswitch-users] api_hangup_hook and bash
Message-ID: <AEEFD410-CEB5-4322-B04D-A7CBA5633FA4@freeswitch.org>

I'm pretty sure that this is doable. Could you give us a hint as to  
what arguments you want to send? For example, do you have one or more  
channel variables you'd like to pass to the shell script?

-MC

Sent from my iPhone

On Dec 23, 2008, at 6:25 PM, Jason White <jason at jasonjgw.net> wrote:

> Frank @ Impact <frank at impactfax.com> wrote:
>> Can this command be used to run a bash script?
>
> Based on information at the wiki, this should be possible; use the  
> system
> command.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From woodydickson at gmail.com  Wed Dec 24 00:04:34 2008
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 24 Dec 2008 16:04:34 +0800
Subject: [Freeswitch-users] Lua script directory
Message-ID: <a44688a0812240004x1a73ea49ne638628ca048900d@mail.gmail.com>

Hi,

Is it possible to change the directory where freeswitch looks for .lua
scripts?
I would like to place the lua scripts in the shared drive so multiple
freeswitch can refer to it.

Thanks,
Woody
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From jaybinks at gmail.com  Wed Dec 24 00:13:15 2008
From: jaybinks at gmail.com (jay binks)
Date: Wed, 24 Dec 2008 18:13:15 +1000
Subject: [Freeswitch-users] Lua script directory
In-Reply-To: <a44688a0812240004x1a73ea49ne638628ca048900d@mail.gmail.com>
References: <a44688a0812240004x1a73ea49ne638628ca048900d@mail.gmail.com>
Message-ID: <ec2ce17e0812240013p56b6cf63oaae2a0ac7143fed9@mail.gmail.com>

use a Dynamic link...

On Wed, Dec 24, 2008 at 6:04 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> Is it possible to change the directory where freeswitch looks for .lua
> scripts?
> I would like to place the lua scripts in the shared drive so multiple
> freeswitch can refer to it.
>
> Thanks,
> Woody
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From yudha2008 at gmail.com  Wed Dec 24 03:23:19 2008
From: yudha2008 at gmail.com (Baskar)
Date: Wed, 24 Dec 2008 16:53:19 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
Message-ID: <d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>

*Hi,

    This is my JavaScript for tone detect

session1 = new Session();
session1.originate(session1, "{ignore_early_media=false}sofia/default/
39841799874 at 172.20.191.228");
session1.execute("tone_detect", "test 400,25 r +1 hangup 'normal_clearing'
1");
session1.execute("bridge", "sofia/default/39841799874 at 172.20.191.228");
session1.execute("transfer", "39841799874");

But Tone Detect does not work at all Did i work wrongly, correct me  where i
am wrong

Another question :

In api tone detect command i got api parameters like this

api tone_detect,Start Tone Detection on a channel,<uuid> <key> <tone_spec>
[<flags> <timeout> <app> <args> <hits>]

In that I want to know what is <tone_spec> ....
I did not see any details in wiki.freeswitch site can any what should be
passed in tone_spec*.

*--
Warm Regards,
N.Baskar*
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From juanbackson at gmail.com  Wed Dec 24 04:40:02 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Wed, 24 Dec 2008 20:40:02 +0800
Subject: [Freeswitch-users] strange error while running stress testing
Message-ID: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>

Hi

I am getting the following strange error while running stress test on
freeswith.  When the number of sessions reaches 3000, I get the
following error:

2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Timeout]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
AUDIO RTP REPORTS ERROR: [Bind Error!]
2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state() RTP Error!
2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
Error[[error near line 1]: root tag missing]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]
2008-12-24 15:37:42 [ERR] sofia_glue.c:490
sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
[Remote Address Error!]

Could someone help me out?  What do those errors mean?

Thanks in advance for all your help.

JB



From msc at freeswitch.org  Wed Dec 24 09:22:36 2008
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Dec 2008 09:22:36 -0800
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
Message-ID: <87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>

On Wed, Dec 24, 2008 at 3:23 AM, Baskar <yudha2008 at gmail.com> wrote:

> *Hi,
>
>     This is my JavaScript for tone detect
>
> session1 = new Session();
> session1.originate(session1, "{ignore_early_media=false}sofia/default/
> 39841799874 at 172.20.191.228");
> session1.execute("tone_detect", "test 400,25 r +1 hangup 'normal_clearing'
> 1");*


Is the combination of 400Hz and 25Hz the correct busy tone for your country?
Also, the +1 means "check for tone for +1 second into the future"
Most likely you want something like +30 so that the tone detect will be
active long enough to hear something!

> *
> session1.execute("bridge", "sofia/default/39841799874 at 172.20.191.228");
> session1.execute("transfer", "39841799874");
>
> But Tone Detect does not work at all Did i work wrongly, correct me  where
> i am wrong
>
> Another question :
>
> In api tone detect command i got api parameters like this
>
> api tone_detect,Start Tone Detection on a channel,<uuid> <key> <tone_spec>
> [<flags> <timeout> <app> <args> <hits>]
>
> In that I want to know what is <tone_spec> ....*


<tone_spec> is talking about the combination of frequencies to listen for.
In your example, the tone spec is "400,25"


Hope that information helped!
-MC (mercutioviz)

*
> I did not see any details in wiki.freeswitch site can any what should be
> passed in tone_spec*.
>
> *--
> Warm Regards,
> N.Baskar*
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Dec 24 10:11:35 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Dec 2008 12:11:35 -0600
Subject: [Freeswitch-users] strange error while running stress testing
In-Reply-To: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>
References: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>
Message-ID: <191c3a030812241011i694d6483u65fe666607686da2@mail.gmail.com>

dont load test against channels that must do a stun lookup.
you are lucky you get 3000 channels doing stun. that's actually an
impressive number.
try it on a lan on a profile with no stun.


On Wed, Dec 24, 2008 at 6:40 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi
>
> I am getting the following strange error while running stress test on
> freeswith.  When the number of sessions reaches 3000, I get the
> following error:
>
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Timeout]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
> AUDIO RTP REPORTS ERROR: [Bind Error!]
> 2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state() RTP
> Error!
> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
> Error[[error near line 1]: root tag missing]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
> [Remote Address Error!]
>
> Could someone help me out?  What do those errors mean?
>
> Thanks in advance for all your help.
>
> JB
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From john at feith.com  Wed Dec 24 13:31:45 2008
From: john at feith.com (John Wehle)
Date: Wed, 24 Dec 2008 16:31:45 -0500 (EST)
Subject: [Freeswitch-users]  another switch_ivr_set_user() can't find user
Message-ID: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>

>>  a) Should sip_auth_realm be set by FreeSWITCH to the value associated
>>     with force-register-domain
>
> You have to remember the default assumes a lot.  You go to changing  
> things you have to then change the way things are assumed.

I appreciate that.  Let me ask the question slightly differently.

sofia_reg_parse_auth contains the following logic:

  if (!switch_strlen_zero(profile->reg_domain)) {
          domain_name = profile->reg_domain;
  } else {
          domain_name = realm;
  }

where profile->reg_domain is set from force-register-domain.
It then calls switch_xml_locate_user using domain_name.
It looks like force-register-domain is intended to make
FreeSWITCH believe that the user is in domain specified by
force-register-domain.

Later there's:

  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
    "sip_auth_realm", realm);
  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
    "domain_name", realm);

Shouldn't the add_header for domain_name contain the value for
the actual domain used to locate the user?

And ideally shouldn't the rest of FreeSWITCH (including examples
intended to get you started) work in the same fashion for consistency
sake (i.e. when trying to locate a user reference the domain used by
sofia_reg_parse_auth to locate the user instead of blindly using
sip_auth_realm)?

My thought is if sofia_reg_parse_auth set things up properly,
then the rest of FreeSWITCH shouldn't know or even care that
force-register-domain is in use ... it should be as if the
VoIP phone had in fact registered using the domain specified
by force-register-domain.

-- John
-------------------------------------------------------------------------
|   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
|    John Wehle    |     Fax: 1-215-540-5495  |                         |
-------------------------------------------------------------------------




From kristjan.ugrin at gmail.com  Wed Dec 24 13:51:41 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Wed, 24 Dec 2008 22:51:41 +0100
Subject: [Freeswitch-users] Behind a router
Message-ID: <op.umoscdqdatcbto@linux.site>

Hello! I installed freeswitch on different configuration, but can't get registration working.
So my phone (192.168.10.2) is now behind a wireless router (192.168.10.1),
ethernet port (192.168.0.5) is attached to eth1 (192.168.0.1).

Fs is running on that machine (it has eth0 - net and eth1 - inside lan), sofia shows:
http://pastebin.com/m6f349c32


and registration fails:
http://pastebin.com/m2916e20d

How could I fix that? So when I would call 1000 at 212.235.180.41 would ring my phone at 192.168.10.2?
I've setup port forwarding on port 5060 between 192.168.10.2 and 192.168.0.1 both ways.


-- 






From brian at freeswitch.org  Wed Dec 24 14:05:26 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Dec 2008 16:05:26 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find user
In-Reply-To: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>
References: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>
Message-ID: <91CBBB52-E51A-44C5-A2AE-EADC5A39560C@freeswitch.org>


On Dec 24, 2008, at 3:31 PM, John Wehle wrote:

>>> a) Should sip_auth_realm be set by FreeSWITCH to the value  
>>> associated
>>>    with force-register-domain
>>
>> You have to remember the default assumes a lot.  You go to changing
>> things you have to then change the way things are assumed.
>
> I appreciate that.  Let me ask the question slightly differently.
>
> sofia_reg_parse_auth contains the following logic:
>
>  if (!switch_strlen_zero(profile->reg_domain)) {
>          domain_name = profile->reg_domain;
>  } else {
>          domain_name = realm;
>  }
>
> where profile->reg_domain is set from force-register-domain.
> It then calls switch_xml_locate_user using domain_name.
> It looks like force-register-domain is intended to make
> FreeSWITCH believe that the user is in domain specified by
> force-register-domain.

Yes that is exactly what that option does. see also force-register-db- 
domain

>
>
> Later there's:
>
>  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
>    "sip_auth_realm", realm);
>  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
>    "domain_name", realm);

This looks like a typo.

> And ideally shouldn't the rest of FreeSWITCH (including examples
> intended to get you started) work in the same fashion for consistency
> sake (i.e. when trying to locate a user reference the domain used by
> sofia_reg_parse_auth to locate the user instead of blindly using
> sip_auth_realm)?

I see it the examples are rather consistent consider its SIP  
centric.   Can you provide more detail?  In FreeSWITCH for the sake of  
sanity the auth_realm is the domain name...

>
>
> My thought is if sofia_reg_parse_auth set things up properly,
> then the rest of FreeSWITCH shouldn't know or even care that
> force-register-domain is in use ... it should be as if the
> VoIP phone had in fact registered using the domain specified
> by force-register-domain.

see force-register-db-domain I think that solves the problem you're  
talking about.

/b

>
>
> -- John
> -------------------------------------------------------------------------
> |   Feith Systems  |   Voice: 1-215-646-8000  |  Email:  
> john at feith.com  |
> |    John Wehle    |     Fax: 1-215-540-5495   
> |                         |
> -------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jason at jasonjgw.net  Wed Dec 24 15:34:51 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 25 Dec 2008 10:34:51 +1100
Subject: [Freeswitch-users] Setting up port audio for incoming/outgoing calls
Message-ID: <20081224233451.GA5687@jdc.jasonjgw.net>

On the wiki, an example of a port audio configuration is given that involves
creating a Sip gateway on localhost. As I couldn't get this to work
(apparently due to the external profile's detection of NAT), I thought I would
try an alternative approach. I am modifying the default dial plan here. At
some point I'll probably just rewrite it anyway.

I have created a user in the directory for extension 1020. For outbound calls,
in default.xml, I have the following:

    <extension name="portaudio" continue="true">
      <condition field="source" expression="mod_portaudio">
        <action application="set_user" data="1020@$${domain}"/>
        <action application="set" data="effective_caller_id_number=1020"/>
      </condition>
    </extension>

The log shows that the set_user is executed, as is the set
effective_caller_id_number (the latter shouldn't be necessary, unless I'm
misunderstanding).

However, running show channels after making a call from the portaudio device
still shows the user name and caller id as
FreeSWITCH,0000000000

Also, when I try to call a local extension from the audio device, I get the
following in the logs, and the call is terminated. I've checked the code, and
clearly the failure to open the file is the cause of the termination. The Sip
phone on the extension rings once and then it receives the cancellation from
FreeSWITCH.

2008-12-25 10:29:59 [DEBUG] switch_ivr_originate.c:1313 switch_ivr_originate()
P
lay Ringback File [local_stream://moh]
2008-12-25 10:29:59 [ERR] mod_local_stream.c:308 local_stream_file_open()
Unknow
n source moh
2008-12-25 10:29:59 [ERR] switch_ivr_originate.c:1322 switch_ivr_originate()
Err
or Playing File
2008-12-25 10:29:59 [DEBUG] switch_core_codec.c:122
switch_core_session_set_read
_codec() Restore original codec.
2008-12-25 10:29:59 [NOTICE] switch_ivr_originate.c:1560
switch_ivr_originate()
Hangup sofia/internal/sip:1000 at 192.168.0.4:2048;line=mxyv04us
[CS_CONSUME_MEDIA]
 [NO_ANSWER]
2008-12-25 10:29:59 [DEBUG] switch_channel.c:1494
switch_channel_perform_hangup(
) Send signal sofia/internal/sip:1000 at 192.168.0.4:2048;line=mxyv04us [KILL]

Any hints would be welcome.  There is no urgency, of course, as I'm doing this
for fun and out of interest.

Happy holidays to all on the FreeSWITCH list.




From krice at suspicious.org  Wed Dec 24 21:12:07 2008
From: krice at suspicious.org (Ken Rice)
Date: Wed, 24 Dec 2008 23:12:07 -0600
Subject: [Freeswitch-users] Happy Holidays
Message-ID: <C5787247.4B6AA%krice@suspicious.org>

Merry Christmas and Chag orim same'ach

Ken





From jason at jasonjgw.net  Wed Dec 24 22:17:38 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 25 Dec 2008 17:17:38 +1100
Subject: [Freeswitch-users] Setting up port audio for
	incoming/outgoing	calls
In-Reply-To: <20081224233451.GA5687@jdc.jasonjgw.net>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
Message-ID: <20081225061738.GA15452@jdc.jasonjgw.net>

I've solved part of my problem.

local_stream_file_open() was looking for moh/48000, because I had set the
sample rate to 48 khz in my portaudio configuration. (The context was that the
music was to be used as ring-back). Not surprisingly, the lookup failed, as
did the lookup for "moh"; if it had been moh/8000 it would have succeeded.

It all makes sense now.




From zolotov at altron.ua  Thu Dec 25 06:49:40 2008
From: zolotov at altron.ua (Evgeniy Zolotov)
Date: Thu, 25 Dec 2008 16:49:40 +0200
Subject: [Freeswitch-users] How PBXs works on different platforms
Message-ID: <1230216580.5361.9.camel@opos20.altron.lan>

  Hello!

Our experience (over 1 month) of FreeSWITCH testing under SunSolaris 10
shows us such things :


1. FreeSWITCH can be built with gcc compiler, which consists at
SunSolaris 10 by default :


$ /usr/sfw/bin/gcc --version

gcc (GCC) 3.4.3 (csl-sol210-3_4-branch+sol_rpath)

Copyright (C) 2004 Free Software Foundation, Inc.

This is free software; see the source for copying conditions. There is
NO

warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR
PURPOSE.


With this compiler is possible built FreeSWITCH except mod_openzap (this
module needs ISO C99 compatible compiler), but, it'll be shown below,
there are no any reasons to make mod_openzap under Solaris 10.


2. Newlyversion of gcc can be taken from CSW ? repositories, for example
here : 


http://mirrors.usc.edu/pub/csw/unstable/i386/5.10/


There are consists all CSW ? packages, which gcc needs for install. Full
list of CSW ? packages in a world can be found here :


http://www.opencsw.org/userguide/


So we have :


$ /opt/csw/gcc4/bin/gcc --version

gcc (GCC) 4.0.2

Copyright (C) 2005 Free Software Foundation, Inc.

This is free software; see the source for copying conditions. There is
NO

warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR
PURPOSE.


>From there we take packages aclocal & automake (they are absent into Sun
Solaris 10 distributive ), if we want build FreeSWITCH ? trunk starting
from ./bootstrap.sh .


3. One more variant of FreeSWITCH installation, which described in a few
sources in the Internet
(http://www.voiceworks.pl/cypromis/category/solaris-opensolaris/) - is
installation of Sun Studio 12 (package
SunStudio12ml-solaris-x86-200709-pkg.tar.bz2 , from Sun Microsystems
official site):


/opt/SUNWspro/bin/cc


Build can be done in a command mode, redefining parameters CC, CCOPT,
LDOPT, or even in Sun Studio GUI IDE, launching it from remote host :


# ssh -nfX <sun-host> sunstudio


4. We built & installed FreeSWITCH + Zaptel + Wanpipe on 2 64-bit
SunSolaris servers


$ isainfo -k

amd64

$ isainfo -v

64-bit amd64 applications

tscp cx16 sse3 sse2 sse fxsr amd_3dnowx amd_3dnow amd_mmx mmx cmov
amd_sysc

cx8 tsc fpu

32-bit i386 applications

tscp cx16 sse3 sse2 sse fxsr amd_3dnowx amd_3dnow amd_mmx mmx cmov
amd_sysc

cx8 tsc fpu


+ on a 64-bit server SuperMicro ? SYS-5025B; proc. Intel Xeon 3210 :


# isainfo -v

64-bit amd64 applications

ssse3 cx16 mon sse3 sse2 sse fxsr mmx cmov amd_sysc cx8 tsc fpu

32-bit i386 applications

ssse3 cx16 mon sse3 sse2 sse fxsr mmx cmov sep cx8 tsc fpu # isainfo -k

amd64


- on a last one we didn't even succeed even to start Zaptel + Wanpipe,
having exactly the same configuration like on previous two.

We didn't succeed to build some of FreeSWITCH libraries with 64-bit
support, but that is a question of time and technology. So we built
32-bit version of FreeSWITCH.


5. This variant was successfully tested for SIP calls and execution of
all extensions, whitch correctly works under Linux.


6. Then we set up support for E1/T1 Sangoma A-104. We found just this :


- ftp://ftp.sangoma.com/Solaris/Beta/SVwanpipe-i386-5.10.pkg

        - ftp://ftp.sangoma.com/Solaris/Packages/SVzaptel-i386-5.10.pkg
        


7. We managed to configure and start that variant (zaptel + TDM Voice
API). Complete equivalent of it successfully worked under Linux CentOS 5
? including configuration FreeSwitch mod_openzap.


# ./wanrouter hwprobe


-------------------------------

| Wanpipe Hardware Probe Info |

-------------------------------

1. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=1 : HWEC=0 : V=33

2. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=2 : HWEC=0 : V=33

3. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=3 : HWEC=0 : V=33

4. AFT-A104-SH : SLOT=2 : BUS=2 : IRQ=5 : CPU=A : PORT=4 : HWEC=0 : V=33


Card Cnt: S508= 0 S514X= 0 S518= 0 A101-2= 0 A104= 1 A300= 0 A200= 0
A108= 0


# ./wanrouter status


Devices currently active:

wanpipe1 wanpipe2 wanpipe3 wanpipe4


Device name | Protocol | Station | Status |

wanpipe4 | AFT TE1 | N/A | Connected |

wanpipe3 | AFT TE1 | N/A | Connected |

wanpipe2 | AFT TE1 | N/A | Connected |

wanpipe1 | AFT TE1 | N/A | Connecting |


I.e. E1 spans 4, 3, 2 connected with cables are perceived by Wanpipe in
a synchronisation mode that is visible also on GREEN LEDs on the card. 


8. But we do not manage to receive any reception PRI events (under
FreeSwitch) that it is possible to receive in precisely same
configuration in Linux.


9.After that we tested just transmition of raw B-channel data with our
own tests, which has been created when we wrote Zaptel drivers for
nonstandart E1 equipment. This tests pass test data sequences into
B-channel and receives them from another B-channel, which connected to
the first B-channel with crosscable; cheks up identity of the received
information and measures delay of data passing on a loop. This tests
works good under Linux (we use them for a year), but under Solaris they
shows us that Zaptel + Wanpipe doesn't receives B-data from B-chans. 

We have source code for SVzaptel-i386-5.10.pkg , developed by SunLabs


https://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk/


Studying of this package and site has shown us such things :

      * SVzaptel-i386-5.10.pkg was developed by little community
        http://www.solarisvoip.com/;
        
      * development of this package moves very slow, last updates ? 2006
        year;
        
      * package has limited functionality (unlike original Zaptel );
        
      * SVzaptel-i386-5.10.pkg was tested on a limited amount of E1/t1
        cards (only one !) - Digium Wildcard TE110P T1/PRI).
        


We doesn't have source code for SVwanpipe- i386-5.10.pkg , so we cann't
say anything about it.


Resume:

Declared, for example: http://en.wikipedia.org/wiki/FreeSWITCH and other
URLs, platform support is realy only for SIP-protocol features but :

      * this can be done without using of any PBX, for example with the
        help of SER/OpenSER or sofia library;
        
      * is grounded that into OS is present : a) C ? compiler; b) NetBSD
        compatible IP-stack (in this case QNX 6.4 and many others OS
        could be included in the list of supported OS)
        


We suppose same things under OS *BSD family.

For objectivity it is necessary to underline that the same state of
affairs takes place and in all others free PBX, realising support E1
through the Zaptel interface: Asterisk, YATE etc. 

And all declarations about ?supported planforms? is only promotional
declarations. 

Though it is tested and affirms at :


http://www.thrallingpenguin.com/articles/asterisk-solaris.htm


- that Solaris releases of PBX should have almost in 2 times the big
productivity, than for Linux ? that has it sense only for narrow case
SIP connections that hold back testers and enter the others into error.

   Any info would be appreciated,
   Evgeniy.







From rehan at supertec.com  Thu Dec 25 05:57:32 2008
From: rehan at supertec.com (Rehan Allah Wala)
Date: Thu, 25 Dec 2008 06:57:32 -0700
Subject: [Freeswitch-users] Happy Holidays
In-Reply-To: <C5787247.4B6AA%krice@suspicious.org>
References: <C5787247.4B6AA%krice@suspicious.org>
Message-ID: <49532EDC.16933.17AC8D5C@rehan.supertec.com>

Thank You for all the wishes people,

Merry Christmas and Happy Holidays to you and every one else .

and a very Happy New Year :)

Rehan

Rehan Ahmed AllahWala
Msn/Yahoo/GoogleTalk/Email: Rehan at Rehan.com
http://www.supertec.com/ - Internet Telephony Solutions
Http://www.DIDX.net - DID Number Market Place.
Don't Remember Me ? Visit http://www.Rehan.com

~~~~~~~~~~~~~~~~~~~
"First they ignore you, then they laugh at you, then they fight you, then you win." 
By Gandhi.

"Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi




From markmorreny at gmail.com  Thu Dec 25 07:41:09 2008
From: markmorreny at gmail.com (mark morreny)
Date: Thu, 25 Dec 2008 23:41:09 +0800
Subject: [Freeswitch-users] what is going on with openmrcp?
Message-ID: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>

Hi,

I checked out Unimrcp and it seems like it is down.  Does anyone know what
is happening to Unimrcp?

It seems like mod_unimrcp is down.

I am looking for a way to integrate Freeswitch to MRCP TTS server.

Is there anywhere to do it?

Thanks,
Mark
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From brian at freeswitch.org  Thu Dec 25 08:45:54 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Dec 2008 10:45:54 -0600
Subject: [Freeswitch-users] Setting up port audio for incoming/outgoing
	calls
In-Reply-To: <20081225061738.GA15452@jdc.jasonjgw.net>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
Message-ID: <EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>

Not quite... you needed to have a moh/48000 defined in localstream  
too. ;)  so when you play local_stream://moh it appends the rate to  
the end to find the exact one.  If you define a moh by itself it would  
have fallen back to that.  See local_stream.conf.xml  ;)

/b

On Dec 25, 2008, at 12:17 AM, Jason White wrote:

> I've solved part of my problem.
>
> local_stream_file_open() was looking for moh/48000, because I had  
> set the
> sample rate to 48 khz in my portaudio configuration. (The context  
> was that the
> music was to be used as ring-back). Not surprisingly, the lookup  
> failed, as
> did the lookup for "moh"; if it had been moh/8000 it would have  
> succeeded.
>
> It all makes sense now.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Thu Dec 25 08:48:14 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Dec 2008 10:48:14 -0600
Subject: [Freeswitch-users] what is going on with openmrcp?
In-Reply-To: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>
References: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>
Message-ID: <023F64F6-B7F0-4A06-8FBB-67B2F4AE909D@freeswitch.org>

Are you confusing openmrcp which is unsupported with unimrcp which  
doesn't have a module for freeswitch yet?

/b

On Dec 25, 2008, at 9:41 AM, mark morreny wrote:

> Hi,
>
> I checked out Unimrcp and it seems like it is down.  Does anyone  
> know what is happening to Unimrcp?
>
> It seems like mod_unimrcp is down.
>
> I am looking for a way to integrate Freeswitch to MRCP TTS server.
>
> Is there anywhere to do it?
>
> Thanks,
> Mark




From andresmartinochoa at gmail.com  Wed Dec 24 23:54:01 2008
From: andresmartinochoa at gmail.com (=?ISO-8859-1?Q?Andr=E9s_Mart=EDn_-_martyn?=)
Date: Thu, 25 Dec 2008 02:54:01 -0500
Subject: [Freeswitch-users] Behind a router
In-Reply-To: <op.umoscdqdatcbto@linux.site>
References: <op.umoscdqdatcbto@linux.site>
Message-ID: <8c1b00b30812242354i49b39ed7u44646a016d817a72@mail.gmail.com>

Hello kriko, I need a question. How do you do for create 1000 account in
freeswitch .. in witch file configuration exacly ? .. I'm beginning with
freewitch. Sorry for not create a new post on forum, but i see this
oportunity for ask you it.

Regards

On Wed, Dec 24, 2008 at 4:51 PM, kriko <kristjan.ugrin at gmail.com> wrote:

> Hello! I installed freeswitch on different configuration, but can't get
> registration working.
> So my phone (192.168.10.2) is now behind a wireless router (192.168.10.1),
> ethernet port (192.168.0.5) is attached to eth1 (192.168.0.1).
>
> Fs is running on that machine (it has eth0 - net and eth1 - inside lan),
> sofia shows:
> http://pastebin.com/m6f349c32
>
>
> and registration fails:
> http://pastebin.com/m2916e20d
>
> How could I fix that? So when I would call 1000 at 212.235.180.41 would ring
> my phone at 192.168.10.2?
> I've setup port forwarding on port 5060 between 192.168.10.2 and
> 192.168.0.1 both ways.
>
>
> --
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Andr?s Mart?n Ochoa;
passport: andresmartin at linuxmail.org;
Linux Registered User #436420;
Asterisk User Number: 1000;
PBX: (57) 1 578 20 30;
Ext: 106
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From andresmartinochoa at gmail.com  Wed Dec 24 23:55:34 2008
From: andresmartinochoa at gmail.com (=?ISO-8859-1?Q?Andr=E9s_Mart=EDn_-_martyn?=)
Date: Thu, 25 Dec 2008 02:55:34 -0500
Subject: [Freeswitch-users] Happy Holidays
In-Reply-To: <C5787247.4B6AA%krice@suspicious.org>
References: <C5787247.4B6AA%krice@suspicious.org>
Message-ID: <8c1b00b30812242355h17254b67nc7f9e460d13c895@mail.gmail.com>

Thank you. The same from Colombia :D

martyn-dev

On Thu, Dec 25, 2008 at 12:12 AM, Ken Rice <krice at suspicious.org> wrote:

> Merry Christmas and Chag orim same'ach
>
> Ken
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Andr?s Mart?n Ochoa;
passport: andresmartin at linuxmail.org;
Linux Registered User #436420;
Asterisk User Number: 1000;
PBX: (57) 1 578 20 30;
Ext: 106
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From mike at jerris.com  Thu Dec 25 09:28:58 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 25 Dec 2008 12:28:58 -0500
Subject: [Freeswitch-users] what is going on with openmrcp?
In-Reply-To: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>
References: <20ad6b920812250741j466c3e79ybe74e6ac05b7e8b7@mail.gmail.com>
Message-ID: <80CE6997-9129-4F88-8E67-8598573AA0AE@jerris.com>


On Dec 25, 2008, at 10:41 AM, mark morreny wrote:

> Hi,
>
> I checked out Unimrcp and it seems like it is down.  Does anyone  
> know what is happening to Unimrcp?
>
> It seems like mod_unimrcp is down.

There is no such thing to my knowledge.

>
> I am looking for a way to integrate Freeswitch to MRCP TTS server.
>
> Is there anywhere to do it?
>

Currently we still support the older library openmrcp with  
mod_openmrcp.  There are thoughts to move to unimrcp in the future.



From can_man at gmx.de  Thu Dec 25 12:38:11 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Thu, 25 Dec 2008 21:38:11 +0100
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
Message-ID: <20081225203811.79240@gmx.net>

Hello,

I am trying to replace some static settings with dynamic ones which are provided by a webserver. I can bridge calls that way, however I just can't get the following transfer to work. 
The transfer works when in public.xml and looks like this:

<extension name="test9">
<condition field="destination_number" expression="^(10001)$">
<action application="set" data="domain_name=$${domain}"/>
<action application="transfer" data="9999 XML default"/>
</condition>
</extension>


The xml received by FS from the webserver looks like this:

<context name="public">
<extension name="test9">
<condition field="destination_number" expression="^(10001)$">
<action application="set" data="domain_name=$${domain}"/>
<action application="transfer" data="9999 XML default"/>
</condition>
</extension>
</context>


I have also tried without context name and extension name, but I got the same result.

This is the console log output:

2008-12-25 21:27:27 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/anonymous at sipgate.de [7281a542-d2c2-11dd-80f0-3fe65955e25b]
2008-12-25 21:27:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing anonymous->10001 in context public
2008-12-25 21:27:29 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML response is in /tmp/72829470-d2c2-11dd-80f0-3fe65955e25b.tmp.xml
2008-12-25 21:27:29 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-25 21:27:29 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 3 (sofia/external/anonymous at sipgate.de) Ended
2008-12-25 21:27:29 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]


The 9999 extension in default looks like this:

    <extension name="hold_music">
      <condition field="destination_number" expression="^9999$"/>
      <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
        <action application="answer"/>
        <action application="execute_extension" data="is_secure XML features"/>
        <action application="playback" data="$${hold_music}"/>
        <anti-action application="answer"/>
        <anti-action application="playback" data="$${hold_music}"/>
      </condition>
    </extension>


Thank you very much for your help.

Phil


-- 
Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL 
f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a



From kristjan.ugrin at gmail.com  Thu Dec 25 13:07:02 2008
From: kristjan.ugrin at gmail.com (kriko)
Date: Thu, 25 Dec 2008 22:07:02 +0100
Subject: [Freeswitch-users] Behind a router
In-Reply-To: <8c1b00b30812242354i49b39ed7u44646a016d817a72@mail.gmail.com>
References: <op.umoscdqdatcbto@linux.site>
	<8c1b00b30812242354i49b39ed7u44646a016d817a72@mail.gmail.com>
Message-ID: <op.umqkxyoeatcbto@tux.tux>

It is already there as a demo. Try to look in config folder, I'm not at my  
workstation atm.

On Thu, 25 Dec 2008 08:54:01 +0100, Andr?s Mart?n - martyn  
<andresmartinochoa at gmail.com> wrote:

> Hello kriko, I need a question. How do you do for create 1000 account in
> freeswitch .. in witch file configuration exacly ? .. I'm beginning with
> freewitch. Sorry for not create a new post on forum, but i see this
> oportunity for ask you it.
>
> Regards
>
> On Wed, Dec 24, 2008 at 4:51 PM, kriko <kristjan.ugrin at gmail.com> wrote:
>
>> Hello! I installed freeswitch on different configuration, but can't get
>> registration working.
>> So my phone (192.168.10.2) is now behind a wireless router  
>> (192.168.10.1),
>> ethernet port (192.168.0.5) is attached to eth1 (192.168.0.1).
>>
>> Fs is running on that machine (it has eth0 - net and eth1 - inside lan),
>> sofia shows:
>> http://pastebin.com/m6f349c32
>>
>>
>> and registration fails:
>> http://pastebin.com/m2916e20d
>>
>> How could I fix that? So when I would call 1000 at 212.235.180.41 would  
>> ring
>> my phone at 192.168.10.2?
>> I've setup port forwarding on port 5060 between 192.168.10.2 and
>> 192.168.0.1 both ways.
>>
>>
>> --
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>



-- 
(\__/)
(='.'=)
(")_(")



From msc at freeswitch.org  Thu Dec 25 14:12:40 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 25 Dec 2008 14:12:40 -0800
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <20081225203811.79240@gmx.net>
References: <20081225203811.79240@gmx.net>
Message-ID: <574EDB32-6665-4675-A651-B8DB58FFCAE9@freeswitch.org>

Phil,
Can you do the same test with debug turned on? F8 or "console loglevel  
debug" will do the trick.

-MC

Sent from my iPhone

On Dec 25, 2008, at 12:38 PM, can_man at gmx.de wrote:

> Hello,
>
> I am trying to replace some static settings with dynamic ones which  
> are provided by a webserver. I can bridge calls that way, however I  
> just can't get the following transfer to work.
> The transfer works when in public.xml and looks like this:
>
> <extension name="test9">
> <condition field="destination_number" expression="^(10001)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="9999 XML default"/>
> </condition>
> </extension>
>
>
> The xml received by FS from the webserver looks like this:
>
> <context name="public">
> <extension name="test9">
> <condition field="destination_number" expression="^(10001)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="9999 XML default"/>
> </condition>
> </extension>
> </context>
>
>
> I have also tried without context name and extension name, but I got  
> the same result.
>
> This is the console log output:
>
> 2008-12-25 21:27:27 [NOTICE] switch_channel.c:565  
> switch_channel_set_name() New Channel sofia/external/anonymous at sipgate.de 
>  [7281a542-d2c2-11dd-80f0-3fe65955e25b]
> 2008-12-25 21:27:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()  
> Processing anonymous->10001 in context public
> 2008-12-25 21:27:29 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML  
> response is in /tmp/72829470-d2c2-11dd-80f0-3fe65955e25b.tmp.xml
> 2008-12-25 21:27:29 [NOTICE] switch_core_state_machine.c:168  
> switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de 
>  [CS_EXECUTE] [NORMAL_CLEARING]
> 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:956  
> switch_core_session_thread() Session 3 (sofia/external/anonymous at sipgate.de 
> ) Ended
> 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:958  
> switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de 
>  [CS_HANGUP]
>
>
> The 9999 extension in default looks like this:
>
>    <extension name="hold_music">
>      <condition field="destination_number" expression="^9999$"/>
>      <condition field="${sip_has_crypto}"  
> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
>        <action application="answer"/>
>        <action application="execute_extension" data="is_secure XML  
> features"/>
>        <action application="playback" data="$${hold_music}"/>
>        <anti-action application="answer"/>
>        <anti-action application="playback" data="$${hold_music}"/>
>      </condition>
>    </extension>
>
>
> Thank you very much for your help.
>
> Phil
>
>
> -- 
> Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL
> f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T456 
> 9a
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Thu Dec 25 14:53:32 2008
From: mike at jerris.com (Michael Jerris)
Date: Thu, 25 Dec 2008 17:53:32 -0500
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <20081225203811.79240@gmx.net>
References: <20081225203811.79240@gmx.net>
Message-ID: <2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>

The $$ substitutions are only done in the static XML files.  Al

On Dec 25, 2008, at 3:38 PM, can_man at gmx.de wrote:

> Hello,
>
> I am trying to replace some static settings with dynamic ones which  
> are provided by a webserver. I can bridge calls that way, however I  
> just can't get the following transfer to work.
> The transfer works when in public.xml and looks like this:
>
> <extension name="test9">
> <condition field="destination_number" expression="^(10001)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="9999 XML default"/>
> </condition>
> </extension>
>
>
> The xml received by FS from the webserver looks like this:
>
> <context name="public">
> <extension name="test9">
> <condition field="destination_number" expression="^(10001)$">
> <action application="set" data="domain_name=$${domain}"/>
> <action application="transfer" data="9999 XML default"/>
> </condition>
> </extension>
> </context>
>
>
> I have also tried without context name and extension name, but I got  
> the same result.
>
> This is the console log output:
>
> 2008-12-25 21:27:27 [NOTICE] switch_channel.c:565  
> switch_channel_set_name() New Channel sofia/external/anonymous at sipgate.de 
>  [7281a542-d2c2-11dd-80f0-3fe65955e25b]
> 2008-12-25 21:27:27 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()  
> Processing anonymous->10001 in context public
> 2008-12-25 21:27:29 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML  
> response is in /tmp/72829470-d2c2-11dd-80f0-3fe65955e25b.tmp.xml
> 2008-12-25 21:27:29 [NOTICE] switch_core_state_machine.c:168  
> switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de 
>  [CS_EXECUTE] [NORMAL_CLEARING]
> 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:956  
> switch_core_session_thread() Session 3 (sofia/external/anonymous at sipgate.de 
> ) Ended
> 2008-12-25 21:27:29 [NOTICE] switch_core_session.c:958  
> switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de 
>  [CS_HANGUP]
>
>
> The 9999 extension in default looks like this:
>
>    <extension name="hold_music">
>      <condition field="destination_number" expression="^9999$"/>
>      <condition field="${sip_has_crypto}"  
> expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
>        <action application="answer"/>
>        <action application="execute_extension" data="is_secure XML  
> features"/>
>        <action application="playback" data="$${hold_music}"/>
>        <anti-action application="answer"/>
>        <anti-action application="playback" data="$${hold_music}"/>
>      </condition>
>    </extension>
>
>
> Thank you very much for your help.
>
> Phil
>
>
> -- 
> Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL
> f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T456 
> 9a
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From can_man at gmx.de  Thu Dec 25 16:01:50 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Fri, 26 Dec 2008 01:01:50 +0100
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>
References: <20081225203811.79240@gmx.net>
	<2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>
Message-ID: <20081226000150.140990@gmx.net>


Hello,

thank you for your answers. I am a step further now, it seems that just the "condition" tags as described in the wiki aren't enough. After sending the following xml I think I get stuck at the point Micheal mentioned:

> The $$ substitutions are only done in the static XML files.  Al

FS complains that: Context default not found


XML:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="RE Dial Plan For FreeSwitch">
<context name="public">
<extension name="test10000">
<condition field="destination_number" expression="^(10000)$">
<action application="set" data="domain_name=192.168.178.22"/>
<action application="transfer" data="9999 XML default"/>
</condition>
</extension>
</context>
</section>
</document>

I also tried: <action application="set" data="domain_name=$${domain}"/>

Any idea what else I could use to make sure the default context is found?


The debug looks like this:

freeswitch at voip> 2008-12-26 00:52:09 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/anonymous at sipgate.de [0acbfb9c-d2df-11dd-9b87-537be0ec7712]
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_NEW
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:375 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State NEW
2008-12-26 00:52:09 [DEBUG] sofia.c:2529 sofia_handle_sip_i_state() Channel sofia/external/anonymous at sipgate.de entering state [received]
2008-12-26 00:52:09 [DEBUG] sofia.c:2533 sofia_handle_sip_i_state() Remote SDP:
v=0
o=root 29259 29259 IN IP4 217.10.67.141
s=session
c=IN IP4 217.10.77.24
t=0 0
m=audio 42748 RTP/AVP 8 0 3 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=direction:active
a=nortpproxy:yes

2008-12-26 00:52:09 [DEBUG] sofia_glue.c:2409 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000]/[PCMU:0:8000]
2008-12-26 00:52:09 [DEBUG] sofia_glue.c:2409 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000]/[PCMA:8:8000]
2008-12-26 00:52:09 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/external/anonymous at sipgate.de PCMA/8000 20 ms 160 samples
2008-12-26 00:52:09 [DEBUG] sofia_glue.c:2373 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101
2008-12-26 00:52:09 [DEBUG] sofia.c:2685 sofia_handle_sip_i_state() (sofia/external/anonymous at sipgate.de) State Change CS_NEW -> CS_INIT
2008-12-26 00:52:09 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_INIT
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State INIT
2008-12-26 00:52:09 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/anonymous at sipgate.de SOFIA INIT
2008-12-26 00:52:09 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/external/anonymous at sipgate.de) State Change CS_INIT -> CS_ROUTING
2008-12-26 00:52:09 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State INIT going to sleep
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_ROUTING
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State ROUTING
2008-12-26 00:52:09 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/external/anonymous at sipgate.de SOFIA ROUTING
2008-12-26 00:52:09 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/external/anonymous at sipgate.de Standard ROUTING
2008-12-26 00:52:09 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing anonymous->10000 in context public
2008-12-26 00:52:11 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML response is in /tmp/0accb9ba-d2df-11dd-9b87-537be0ec7712.tmp.xml
2008-12-26 00:52:11 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [test10000] destination_number(10000) =~ /^(10000)$/
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/external/anonymous at sipgate.de) State Change CS_ROUTING -> CS_EXECUTE
2008-12-26 00:52:11 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State ROUTING going to sleep
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_EXECUTE
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE
2008-12-26 00:52:11 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/external/anonymous at sipgate.de SOFIA EXECUTE
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:137 switch_core_standard_on_execute() sofia/external/anonymous at sipgate.de Standard EXECUTE
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/external/anonymous at sipgate.de Execute set(domain_name=192.168.178.22)
2008-12-26 00:52:11 [DEBUG] mod_dptools.c:681 set_function() sofia/external/anonymous at sipgate.de SET [domain_name]=[192.168.178.22]
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/external/anonymous at sipgate.de Execute transfer(9999 XML default)
2008-12-26 00:52:11 [DEBUG] switch_ivr.c:1245 switch_ivr_session_transfer() (sofia/external/anonymous at sipgate.de) State Change CS_EXECUTE -> CS_ROUTING
2008-12-26 00:52:11 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:11 [DEBUG] switch_ivr.c:1249 switch_ivr_session_transfer() sofia/external/anonymous at sipgate.de receive message [SWITCH_MESSAGE_INDICATE_TRANSFER]
2008-12-26 00:52:11 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:11 [NOTICE] switch_ivr.c:1251 switch_ivr_session_transfer() Transfer sofia/external/anonymous at sipgate.de to XML[9999 at default]
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE going to sleep
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_ROUTING
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State ROUTING
2008-12-26 00:52:11 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/external/anonymous at sipgate.de SOFIA ROUTING
2008-12-26 00:52:11 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/external/anonymous at sipgate.de Standard ROUTING
2008-12-26 00:52:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing anonymous->9999 in context default
2008-12-26 00:52:13 [CONSOLE] mod_xml_curl.c:236 xml_url_fetch() XML response is in /tmp/0c0cade4-d2df-11dd-9b87-537be0ec7712.tmp.xml
2008-12-26 00:52:13 [WARNING] mod_dialplan_xml.c:263 dialplan_hunt() Context default not found
2008-12-26 00:52:13 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting
2008-12-26 00:52:13 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup sofia/external/anonymous at sipgate.de [CS_ROUTING] [NO_ROUTE_DESTINATION]
2008-12-26 00:52:13 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de [KILL]
2008-12-26 00:52:13 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:435 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State ROUTING going to sleep
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_HANGUP
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP
2008-12-26 00:52:13 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous at sipgate.de hanging up, cause: NO_ROUTE_DESTINATION
2008-12-26 00:52:13 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 404
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/anonymous at sipgate.de Standard HANGUP, cause: NO_ROUTE_DESTINATION
2008-12-26 00:52:13 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP going to sleep
2008-12-26 00:52:13 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 22 (sofia/external/anonymous at sipgate.de) Locked, Waiting on external entities
2008-12-26 00:52:13 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 22 (sofia/external/anonymous at sipgate.de) Ended
2008-12-26 00:52:13 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]


Thank you.

Phil

> On Dec 25, 2008, at 3:38 PM, can_man at gmx.de wrote:
> 
> > Hello,
> >
> > I am trying to replace some static settings with dynamic ones which  
> > are provided by a webserver. I can bridge calls that way, however I  
> > just can't get the following transfer to work.
> >
> >
> > The 9999 extension in default looks like this:
> >
> >    <extension name="hold_music">
> >      <condition field="destination_number" expression="^9999$"/>
> >      <condition field="${sip_has_crypto}"  
> > expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
> >        <action application="answer"/>
> >        <action application="execute_extension" data="is_secure XML  
> > features"/>
> >        <action application="playback" data="$${hold_music}"/>
> >        <anti-action application="answer"/>
> >        <anti-action application="playback" data="$${hold_music}"/>
> >      </condition>
> >    </extension>
> >
> >



-- 
Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL 
f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a



From jason at jasonjgw.net  Thu Dec 25 16:12:21 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 26 Dec 2008 11:12:21 +1100
Subject: [Freeswitch-users] Setting up port audio for
	incoming/outgoing	calls
In-Reply-To: <EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
	<EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
Message-ID: <20081226001221.GA5201@jdc.jasonjgw.net>

Thanks are due to Brian for his help with this.

Now, how do I set up my configuration for outgoing calls so that, when I make
a call from the portaudio module, the caller_id_number and caller_id_name will
be stored in the database as the extension I want, rather than as 0000000000,
FreeSWITCH?

I've tried setting origination_caller_id_name, origination_caller_id_number,
caller_id_name and caller_id_number in the dial plan, but obviously it enters
the database before these changes take effect.

Thanks to FreeSWITCH developers and other community members for your patience
and helpfulness with these inquiries, and feel free to delay your replies
until after the holidays if desired.




From brian at freeswitch.org  Thu Dec 25 16:51:16 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Dec 2008 18:51:16 -0600
Subject: [Freeswitch-users] Setting up port audio for incoming/outgoing
	calls
In-Reply-To: <20081226001221.GA5201@jdc.jasonjgw.net>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
	<EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
	<20081226001221.GA5201@jdc.jasonjgw.net>
Message-ID: <16448C42-9DDA-4063-B881-895C4BCCB873@freeswitch.org>

Open up portaudi.conf.xml and look for the callerid settings.

/b

On Dec 25, 2008, at 6:12 PM, Jason White wrote:

> Now, how do I set up my configuration for outgoing calls so that,  
> when I make
> a call from the portaudio module, the caller_id_number and  
> caller_id_name will
> be stored in the database as the extension I want, rather than as  
> 0000000000,
> FreeSWITCH?




From jason at jasonjgw.net  Thu Dec 25 17:04:47 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 26 Dec 2008 12:04:47 +1100
Subject: [Freeswitch-users] Setting up port audio for
	incoming/outgoing	calls
In-Reply-To: <16448C42-9DDA-4063-B881-895C4BCCB873@freeswitch.org>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
	<EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
	<20081226001221.GA5201@jdc.jasonjgw.net>
	<16448C42-9DDA-4063-B881-895C4BCCB873@freeswitch.org>
Message-ID: <20081226010447.GA6037@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> wrote:
> Open up portaudi.conf.xml and look for the callerid settings.

How could I possibly have missed that?

Sorry!





From brian at freeswitch.org  Thu Dec 25 17:19:25 2008
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Dec 2008 19:19:25 -0600
Subject: [Freeswitch-users] Setting up port audio for incoming/outgoing
	calls
In-Reply-To: <20081226010447.GA6037@jdc.jasonjgw.net>
References: <20081224233451.GA5687@jdc.jasonjgw.net>
	<20081225061738.GA15452@jdc.jasonjgw.net>
	<EA0FFE19-6634-40CA-847B-5791A16148A0@freeswitch.org>
	<20081226001221.GA5201@jdc.jasonjgw.net>
	<16448C42-9DDA-4063-B881-895C4BCCB873@freeswitch.org>
	<20081226010447.GA6037@jdc.jasonjgw.net>
Message-ID: <FF32E0F3-C1A5-4417-9B62-B46D4617889D@freeswitch.org>

:P  happens to the best of us.

/b

On Dec 25, 2008, at 7:04 PM, Jason White wrote:

> How could I possibly have missed that?
>
> Sorry!




From markmorreny at gmail.com  Thu Dec 25 19:32:39 2008
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 26 Dec 2008 11:32:39 +0800
Subject: [Freeswitch-users] Limiting port for OpenMRCP
Message-ID: <20ad6b920812251932md7bc1dfq9cedbe61070418d8@mail.gmail.com>

Hi,

I have a dev version of a MRCP-supported TTS, but it can only allow 1 port
to connect.  Does OpenMRCP has a way to limit the number of port to connect
and queue the rest of the request somehow?

Thanks for your suggestion and help.

Mark
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From simon0922 at gmail.com  Fri Dec 26 09:22:20 2008
From: simon0922 at gmail.com (Simon Leck)
Date: Fri, 26 Dec 2008 09:22:20 -0800
Subject: [Freeswitch-users] NAT Help needed
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAAZE0eGOLxxMmp0+fzQ6K2HCgAAAEAAAAA+x9JO5/VNPm/XlGkrUG1oBAAAAAA==@gmail.com>

Hi Everybody I am a newbie in this. I have managed to setup freeSwitch but I
am unable to resolve the NAT issue. Hope somebody out there can furnish me
with guidance.

 

So far I have manage to use "Use Stun server to resolve registration problem
regarding client behind NAT device but behind two layers, NAT devices still
have problem in registration.

 

Thanks Everybody for your kind assistance.

 

Simon

Email: simon0922 at gmail.com

 

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From jason at jasonjgw.net  Thu Dec 25 21:17:51 2008
From: jason at jasonjgw.net (Jason White)
Date: Fri, 26 Dec 2008 16:17:51 +1100
Subject: [Freeswitch-users] NAT Help needed
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAAZE0eGOLxxMmp0+fzQ6K2HCgAAAEAAAAA+x9JO5/VNPm/XlGkrUG1oBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAAZE0eGOLxxMmp0+fzQ6K2HCgAAAEAAAAA+x9JO5/VNPm/XlGkrUG1oBAAAAAA==@gmail.com>
Message-ID: <20081226051751.GA17995@jdc.jasonjgw.net>

I'm no NAT expert, but some NAT devices can be configured to translate the
addresses in SIP messages appropriately during routing.

Linux IPTables has a SIP conntrack option (according to my Web search, it's in
kernels >= 2.6.18) that does this.

cisco IOS also supports it - I'm not sure when that feature was introduced.




From markmorreny at gmail.com  Fri Dec 26 00:49:34 2008
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 26 Dec 2008 16:49:34 +0800
Subject: [Freeswitch-users] Need help with openmrcp setup
Message-ID: <20ad6b920812260049x28d6ad7bx4b492b06fc8ecc99@mail.gmail.com>

Hi

I tried to setup mod_openmrcp according to wiki, but I am getting the
following error:

2008-12-27 00:40:23 [ERR] switch_core_speech.c:60 switch_core_speech_open()
Invalid speech module [openmrcp]!
2008-12-27 00:40:23 [ERR] switch_ivr_play_say.c:1848 switch_ivr_speak_text()
Invalid TTS module!

Here is my dialplan

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="voicemail dialplan">
<context name="public">
<extension name="test9">
<condition field="destination_number" expression="^(.*)$">
<action application="speak" data="openmrcp:openmrcp-v2|Susan|Hello World"/>
</condition>
</extension>
</context>
</section>
</document>


I checked that mod_openmrcp.so is compiled ok:

[root at localhost bin]# ls ../mod/mod_openmrcp.so -al
-rwxr-xr-x 1 root root 4981560 Dec 26 19:56 ../mod/mod_openmrcp.so
[root at localhost bin]#

Here is what I have in the openmrcp xml config:

[root at localhost freeswitch]# cat conf/mrcp_profiles/openmrcp-v2.xml
<include>
  <profile name="openmrcp-v2" description="OpenMRCPv2 Server">
    <param name="proto_version" value="2"/>

    <param name="client_ip" value="127.0.0.1"/>
    <param name="client_port" value="8062"/>

    <param name="server_ip" value="192.168.1.109"/>
    <param name="server_port" value="5060"/>

    <param name="rtp_port_min" value="11000"/>
    <param name="rtp_port_max" value="12000"/>
  </profile>
</include>

Could someone help me out?  I would greatly appreciate any help.

Thanks,
Mark
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From yudha2008 at gmail.com  Fri Dec 26 03:16:44 2008
From: yudha2008 at gmail.com (Baskar)
Date: Fri, 26 Dec 2008 16:46:44 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
Message-ID: <d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>

*Hi Michael,
*
    * I have updated all the changes what u said, But still i did not get
any tone detect in the script *

*session1 = new Session();
session1.originate(session1, "{ignore_early_media=True}sofia/default/
39841799874 at 172.20.191.228");
session1.execute("tone_detect", "test 400 r +30 hangup 'normal_clearing'
3");
session1.execute("bridge", "sofia/default/39841799874 at 172.20.191.228");
session1.execute("transfer", "39841799874");

Still i did not know what is the error in the script or some thing
else.please guide me to run this tone detect process.

I have tried through api command also i did not get any update.
I did not get any error also but the tone detect process is not working.

I have tried some combination's like this*

*api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30
transfer '1007 XML default' 3 *
*Output:*
Content-Type: api/response
Content-Length: 45
+OK Enabling tone detection 'test' '400' 'r'

*api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30
transfer '1007' 3*
*Output:*
Content-Type: api/response
Content-Length: 45
+OK Enabling tone detection 'test' '400' 'r'


*api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30
transfer '1007 at 172.20.201.67' 3*
*Output:*
Content-Type: api/response
Content-Length: 45
+OK Enabling tone detection 'test' '400' 'r'
*
api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +1 hangup
'normal_clearing' 3*
*Output:*
Content-Type: api/response
Content-Length: 45
+OK Enabling tone detection 'test' '400' 'r'

*I get output == OK but no detect of tone.*

*"Is there any modules for tone detect to be enabled" please assist me, so
that it is useful for me!*

*Thanks in Advance.*
    *
Warm Regards,
N.Baskar
*
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From juanbackson at gmail.com  Fri Dec 26 03:33:50 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Fri, 26 Dec 2008 19:33:50 +0800
Subject: [Freeswitch-users] Hard limit on RTP sessions
Message-ID: <27c25bc40812260333o4067813bmdc6a63e1af4ffd6b@mail.gmail.com>

Hi,

Is there any hard limit set on the number of RTP sessions for
Freeswitch?  I am seeing freeswitch start sending out BYE after the
number of RTP session reaches 3000.   This problem happens even when
the machine utilization is still low.

Does anyone know what is wrong?

Thanks,
JB



From juanbackson at gmail.com  Fri Dec 26 04:23:33 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Fri, 26 Dec 2008 20:23:33 +0800
Subject: [Freeswitch-users] strange error while running stress testing
In-Reply-To: <191c3a030812241011i694d6483u65fe666607686da2@mail.gmail.com>
References: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>
	<191c3a030812241011i694d6483u65fe666607686da2@mail.gmail.com>
Message-ID: <27c25bc40812260423t22da28d9y56cc7071689b1613@mail.gmail.com>

Hi

The strange thing is that I am not using stun at all.  Also, this stun
error only pops up after the number of sessions reach a certain
amount.

JB

On Thu, Dec 25, 2008 at 2:11 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> dont load test against channels that must do a stun lookup.
> you are lucky you get 3000 channels doing stun. that's actually an
> impressive number.
> try it on a lan on a profile with no stun.
>
>
> On Wed, Dec 24, 2008 at 6:40 AM, Juan Backson <juanbackson at gmail.com> wrote:
>>
>> Hi
>>
>> I am getting the following strange error while running stress test on
>> freeswith.  When the number of sessions reaches 3000, I get the
>> following error:
>>
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Timeout]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp()
>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>> 2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state() RTP
>> Error!
>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>> Error[[error near line 1]: root tag missing]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478
>> [Remote Address Error!]
>>
>> Could someone help me out?  What do those errors mean?
>>
>> Thanks in advance for all your help.
>>
>> JB
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mike at jerris.com  Fri Dec 26 05:31:26 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 08:31:26 -0500
Subject: [Freeswitch-users] Need help with openmrcp setup
In-Reply-To: <20ad6b920812260049x28d6ad7bx4b492b06fc8ecc99@mail.gmail.com>
References: <20ad6b920812260049x28d6ad7bx4b492b06fc8ecc99@mail.gmail.com>
Message-ID: <8C88DEE7-36F6-40E6-9C17-19C22AF825F1@jerris.com>

It looks like mod_openmrcp isn't loaded.


On Dec 26, 2008, at 3:49 AM, mark morreny wrote:

> Hi
>
> I tried to setup mod_openmrcp according to wiki, but I am getting  
> the following error:
>
> 2008-12-27 00:40:23 [ERR] switch_core_speech.c:60  
> switch_core_speech_open() Invalid speech module [openmrcp]!
> 2008-12-27 00:40:23 [ERR] switch_ivr_play_say.c:1848  
> switch_ivr_speak_text() Invalid TTS module!
>
> Here is my dialplan
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="voicemail dialplan">
> <context name="public">
> <extension name="test9">
> <condition field="destination_number" expression="^(.*)$">
> <action application="speak" data="openmrcp:openmrcp-v2|Susan|Hello  
> World"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
>
> I checked that mod_openmrcp.so is compiled ok:
>
> [root at localhost bin]# ls ../mod/mod_openmrcp.so -al
> -rwxr-xr-x 1 root root 4981560 Dec 26 19:56 ../mod/mod_openmrcp.so
> [root at localhost bin]#
>
> Here is what I have in the openmrcp xml config:
>
> [root at localhost freeswitch]# cat conf/mrcp_profiles/openmrcp-v2.xml
> <include>
>   <profile name="openmrcp-v2" description="OpenMRCPv2 Server">
>     <param name="proto_version" value="2"/>
>
>     <param name="client_ip" value="127.0.0.1"/>
>     <param name="client_port" value="8062"/>
>
>     <param name="server_ip" value="192.168.1.109"/>
>     <param name="server_port" value="5060"/>
>
>     <param name="rtp_port_min" value="11000"/>
>     <param name="rtp_port_max" value="12000"/>
>   </profile>
> </include>
>
> Could someone help me out?  I would greatly appreciate any help.
>
> Thanks,
> Mark
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Dec 26 05:32:54 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 08:32:54 -0500
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<2ea4d47e0810160131s40157cbh679d6fc84f13b3d7@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
Message-ID: <BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>

Your still ignoring early media.  Are you trying to detect the tone  
before or after answer?

Mike

On Dec 26, 2008, at 6:16 AM, Baskar wrote:

> Hi Michael,
>
>      I have updated all the changes what u said, But still i did not  
> get any tone detect in the script
>
> session1 = new Session();
> session1.originate(session1, "{ignore_early_media=True}sofia/default/39841799874 at 172.20.191.228 
> ");
> session1.execute("tone_detect", "test 400 r +30 hangup  
> 'normal_clearing' 3");
> session1.execute("bridge", "sofia/default/ 
> 39841799874 at 172.20.191.228");
> session1.execute("transfer", "39841799874");
>
> Still i did not know what is the error in the script or some thing  
> else.please guide me to run this tone detect process.
>
> I have tried through api command also i did not get any update.
> I did not get any error also but the tone detect process is not  
> working.
>
> I have tried some combination's like this
>
> api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30  
> transfer '1007 XML default' 3
> Output:
> Content-Type: api/response
> Content-Length: 45
> +OK Enabling tone detection 'test' '400' 'r'
>
> api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30  
> transfer '1007' 3
> Output:
> Content-Type: api/response
> Content-Length: 45
> +OK Enabling tone detection 'test' '400' 'r'
>
>
> api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +30  
> transfer '1007 at 172.20.201.67' 3
> Output:
> Content-Type: api/response
> Content-Length: 45
> +OK Enabling tone detection 'test' '400' 'r'
>
> api tone_detect 0b2eec58-cbb3-484e-bc60-e274a74337ca busy 400 r +1  
> hangup 'normal_clearing' 3
> Output:
> Content-Type: api/response
> Content-Length: 45
> +OK Enabling tone detection 'test' '400' 'r'
>
> I get output == OK but no detect of tone.
>
> "Is there any modules for tone detect to be enabled" please assist  
> me, so that it is useful for me!
>
> Thanks in Advance.
>
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Dec 26 05:34:56 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 08:34:56 -0500
Subject: [Freeswitch-users] strange error while running stress testing
In-Reply-To: <27c25bc40812260423t22da28d9y56cc7071689b1613@mail.gmail.com>
References: <27c25bc40812240440h61d47d1eob26b24214d8179bf@mail.gmail.com>
	<191c3a030812241011i694d6483u65fe666607686da2@mail.gmail.com>
	<27c25bc40812260423t22da28d9y56cc7071689b1613@mail.gmail.com>
Message-ID: <53316117-BD3B-4B8D-81A7-6FBEF61B0877@jerris.com>

You are indeed using stun.  Check vars.xml if your using the default  
configs.  Also there is config params for the rtp port range,

Mike

On Dec 26, 2008, at 7:23 AM, Juan Backson wrote:

> Hi
>
> The strange thing is that I am not using stun at all.  Also, this stun
> error only pops up after the number of sessions reach a certain
> amount.
>
> JB
>
> On Thu, Dec 25, 2008 at 2:11 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> dont load test against channels that must do a stun lookup.
>> you are lucky you get 3000 channels doing stun. that's actually an
>> impressive number.
>> try it on a lan on a profile with no stun.
>>
>>
>> On Wed, Dec 24, 2008 at 6:40 AM, Juan Backson  
>> <juanbackson at gmail.com> wrote:
>>>
>>> Hi
>>>
>>> I am getting the following strange error while running stress test  
>>> on
>>> freeswith.  When the number of sessions reaches 3000, I get the
>>> following error:
>>>
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Timeout]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:2056  
>>> sofia_glue_activate_rtp()
>>> AUDIO RTP REPORTS ERROR: [Bind Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state()  
>>> RTP
>>> Error!
>>> 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate()
>>> Error[[error near line 1]: root tag missing]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>> 2008-12-24 15:37:42 [ERR] sofia_glue.c:490
>>> sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: 
>>> 3478
>>> [Remote Address Error!]
>>>
>>> Could someone help me out?  What do those errors mean?
>>>
>>> Thanks in advance for all your help.
>>>
>>> JB
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From markmorreny at gmail.com  Fri Dec 26 06:08:34 2008
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 26 Dec 2008 22:08:34 +0800
Subject: [Freeswitch-users] Need help with openmrcp setup
In-Reply-To: <8C88DEE7-36F6-40E6-9C17-19C22AF825F1@jerris.com>
References: <20ad6b920812260049x28d6ad7bx4b492b06fc8ecc99@mail.gmail.com>
	<8C88DEE7-36F6-40E6-9C17-19C22AF825F1@jerris.com>
Message-ID: <20ad6b920812260608h19646e8w212eca4a249db03@mail.gmail.com>

Hi,

Thanks for the hint.  I checked the log again and found out:

2008-12-27 06:01:49 [NOTICE] switch_loadable_module.c:259
switch_loadable_module_process() Adding API Function 'lua'
2008-12-27 06:01:49 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_openmrcp.so
**/usr/local/freeswitch/mod/mod_openmrcp.so: undefined symbol:
TLSv1_method**

What is causing this problem?

Thanks for all your help.

Mark

On Fri, Dec 26, 2008 at 9:31 PM, Michael Jerris <mike at jerris.com> wrote:

> It looks like mod_openmrcp isn't loaded.
>
> On Dec 26, 2008, at 3:49 AM, mark morreny wrote:
>
> Hi
>
> I tried to setup mod_openmrcp according to wiki, but I am getting the
> following error:
>
> 2008-12-27 00:40:23 [ERR] switch_core_speech.c:60 switch_core_speech_open()
> Invalid speech module [openmrcp]!
> 2008-12-27 00:40:23 [ERR] switch_ivr_play_say.c:1848
> switch_ivr_speak_text() Invalid TTS module!
>
> Here is my dialplan
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="voicemail dialplan">
> <context name="public">
> <extension name="test9">
> <condition field="destination_number" expression="^(.*)$">
> <action application="speak" data="openmrcp:openmrcp-v2|Susan|Hello World"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
>
> I checked that mod_openmrcp.so is compiled ok:
>
> [root at localhost bin]# ls ../mod/mod_openmrcp.so -al
> -rwxr-xr-x 1 root root 4981560 Dec 26 19:56 ../mod/mod_openmrcp.so
> [root at localhost bin]#
>
> Here is what I have in the openmrcp xml config:
>
> [root at localhost freeswitch]# cat conf/mrcp_profiles/openmrcp-v2.xml
> <include>
>   <profile name="openmrcp-v2" description="OpenMRCPv2 Server">
>     <param name="proto_version" value="2"/>
>
>     <param name="client_ip" value="127.0.0.1"/>
>     <param name="client_port" value="8062"/>
>
>     <param name="server_ip" value="192.168.1.109"/>
>     <param name="server_port" value="5060"/>
>
>     <param name="rtp_port_min" value="11000"/>
>     <param name="rtp_port_max" value="12000"/>
>   </profile>
> </include>
>
> Could someone help me out?  I would greatly appreciate any help.
>
> Thanks,
> Mark
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Dec 26 08:06:03 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Dec 2008 10:06:03 -0600
Subject: [Freeswitch-users] another switch_ivr_set_user() can't find user
In-Reply-To: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>
References: <200812242131.mBOLVjNc006911@jwlab.FEITH.COM>
Message-ID: <191c3a030812260806m2c8739eew231f832354d3aa15@mail.gmail.com>

If anything should be changed it's to add an additional
actual_register_domain header in the cases when it's being forced but it's
not completely necessary.

Typical example is when a client is using the ip address in the domain field
and you want to force it
to point at the domain name in your registry.

The point of the param is to ignore the real domain supplied in the client
and normalize all registrations to a certian domain in your db.
if you want to call registered users with this mode of operation you also
need the force-register-db-domain
which takes it a step further and writes the forced domain into the
registration db so when you try to call user/<id>@<domain>
it will find it.




On Wed, Dec 24, 2008 at 3:31 PM, John Wehle <john at feith.com> wrote:

> >>  a) Should sip_auth_realm be set by FreeSWITCH to the value associated
> >>     with force-register-domain
> >
> > You have to remember the default assumes a lot.  You go to changing
> > things you have to then change the way things are assumed.
>
> I appreciate that.  Let me ask the question slightly differently.
>
> sofia_reg_parse_auth contains the following logic:
>
>  if (!switch_strlen_zero(profile->reg_domain)) {
>          domain_name = profile->reg_domain;
>  } else {
>          domain_name = realm;
>  }
>
> where profile->reg_domain is set from force-register-domain.
> It then calls switch_xml_locate_user using domain_name.
> It looks like force-register-domain is intended to make
> FreeSWITCH believe that the user is in domain specified by
> force-register-domain.
>
> Later there's:
>
>  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
>    "sip_auth_realm", realm);
>  switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM,
>    "domain_name", realm);
>
> Shouldn't the add_header for domain_name contain the value for
> the actual domain used to locate the user?
>
> And ideally shouldn't the rest of FreeSWITCH (including examples
> intended to get you started) work in the same fashion for consistency
> sake (i.e. when trying to locate a user reference the domain used by
> sofia_reg_parse_auth to locate the user instead of blindly using
> sip_auth_realm)?
>
> My thought is if sofia_reg_parse_auth set things up properly,
> then the rest of FreeSWITCH shouldn't know or even care that
> force-register-domain is in use ... it should be as if the
> VoIP phone had in fact registered using the domain specified
> by force-register-domain.
>
> -- John
> -------------------------------------------------------------------------
> |   Feith Systems  |   Voice: 1-215-646-8000  |  Email: john at feith.com  |
> |    John Wehle    |     Fax: 1-215-540-5495  |                         |
> -------------------------------------------------------------------------
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Dec 26 08:11:07 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Dec 2008 10:11:07 -0600
Subject: [Freeswitch-users] Hard limit on RTP sessions
In-Reply-To: <27c25bc40812260333o4067813bmdc6a63e1af4ffd6b@mail.gmail.com>
References: <27c25bc40812260333o4067813bmdc6a63e1af4ffd6b@mail.gmail.com>
Message-ID: <191c3a030812260811x56def967lc8a68755f62b7e54@mail.gmail.com>

I think what's wrong is that instead of listening to the explanation on the
other thread you already started on this issue, you started a new thread
trying to ask the same question a different way hoping for a different
answer.  Once you learn how to configure FreeSWITCH you will be able to stop
using the profile with stun enabled for your testing.


On Fri, Dec 26, 2008 at 5:33 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> Is there any hard limit set on the number of RTP sessions for
> Freeswitch?  I am seeing freeswitch start sending out BYE after the
> number of RTP session reaches 3000.   This problem happens even when
> the machine utilization is still low.
>
> Does anyone know what is wrong?
>
> Thanks,
> JB
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From adnan at barakatdesigns.net  Fri Dec 26 08:19:51 2008
From: adnan at barakatdesigns.net (Adnan Barakat)
Date: Fri, 26 Dec 2008 16:19:51 +0000
Subject: [Freeswitch-users] group_confirm seems to be broken
Message-ID: <49550427.70606@barakatdesigns.net>

Hi all,

I've just updated from r10000 to the latest trunk (as I needed 
mod_http), and group_confirm seems to have broken after the update. Now 
the first 2-5 seconds of the file is played very quickly at poor 
quality, then the end of the file plays fine.

Here is the relevant part of the dialplan;
<action application="set" data="timer_name=soft"/>
<action application="set" data="group_confirm_key=exec"/>
<action application="set" data="group_confirm_file=playback /somefile.al"/>


The last time this happened timer_name=soft fixed the problem, but 
doesn't seem to make any difference in this case.


Thanks

Adnan



From frank at impactfax.com  Fri Dec 26 08:30:35 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 26 Dec 2008 11:30:35 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <AEEFD410-CEB5-4322-B04D-A7CBA5633FA4@freeswitch.org>
Message-ID: <565001c96777$47a318d0$33014c0a@ws4>

All I am passing into the script is the recording file name.

I tried using the system command right after the bridge command but
before a hangup command.  Thusly,

<action application="bridge" data="${enum_auto_route}"/>
<action application="system" data="/root/procrecording.sh
$${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
<action application="hangup"/>

The problem I am seeing is that sometimes this script gets run and
sometimes it does not.  I think it has to do maybe with which end hangs
up the phone.  But I cannot seem to nail it down just yet...

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins
 
I'm pretty sure that this is doable. Could you give us a hint as to  
what arguments you want to send? For example, do you have one or more  
channel variables you'd like to pass to the shell script?

-MC





From frank at impactfax.com  Fri Dec 26 08:57:15 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 26 Dec 2008 11:57:15 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <565001c96777$47a318d0$33014c0a@ws4>
Message-ID: <567c01c9677b$01739480$33014c0a@ws4>

I have confirmed that this system call does not fire if the calling
party hangs up the phone first.  Is there a way to get the script to
fire regardless of who hangs up first?

-F

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Frank @ Impact

<action application="bridge" data="${enum_auto_route}"/>
<action application="system" data="/root/procrecording.sh
$${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
<action application="hangup"/>

The problem I am seeing is that sometimes this script gets run and
sometimes it does not.  I think it has to do maybe with which end hangs
up the phone.  But I cannot seem to nail it down just yet...





From brian at freeswitch.org  Fri Dec 26 09:09:22 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Dec 2008 11:09:22 -0600
Subject: [Freeswitch-users] group_confirm seems to be broken
In-Reply-To: <49550427.70606@barakatdesigns.net>
References: <49550427.70606@barakatdesigns.net>
Message-ID: <EE30E7F5-1E96-4280-9107-5D12B3136FD7@freeswitch.org>

Please update and try again.

Committed revision 10949.

/b

On Dec 26, 2008, at 10:19 AM, Adnan Barakat wrote:

> Hi all,
>
> I've just updated from r10000 to the latest trunk (as I needed
> mod_http), and group_confirm seems to have broken after the update.  
> Now
> the first 2-5 seconds of the file is played very quickly at poor
> quality, then the end of the file plays fine.
>
> Here is the relevant part of the dialplan;
> <action application="set" data="timer_name=soft"/>
> <action application="set" data="group_confirm_key=exec"/>
> <action application="set" data="group_confirm_file=playback / 
> somefile.al"/>
>
>
> The last time this happened timer_name=soft fixed the problem, but
> doesn't seem to make any difference in this case.
>
>
> Thanks
>
> Adnan
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From adnan at barakatdesigns.net  Fri Dec 26 09:38:03 2008
From: adnan at barakatdesigns.net (Adnan Barakat)
Date: Fri, 26 Dec 2008 17:38:03 +0000
Subject: [Freeswitch-users] group_confirm seems to be broken
In-Reply-To: <EE30E7F5-1E96-4280-9107-5D12B3136FD7@freeswitch.org>
References: <49550427.70606@barakatdesigns.net>
	<EE30E7F5-1E96-4280-9107-5D12B3136FD7@freeswitch.org>
Message-ID: <4955167B.7070107@barakatdesigns.net>

Brian West wrote:
> Please update and try again.
> 
> Committed revision 10949.
Thanks Brian, works perfectly.


Adnan



From brian at freeswitch.org  Fri Dec 26 09:46:11 2008
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Dec 2008 11:46:11 -0600
Subject: [Freeswitch-users] group_confirm seems to be broken
In-Reply-To: <4955167B.7070107@barakatdesigns.net>
References: <49550427.70606@barakatdesigns.net>
	<EE30E7F5-1E96-4280-9107-5D12B3136FD7@freeswitch.org>
	<4955167B.7070107@barakatdesigns.net>
Message-ID: <DDDB5D15-027F-44C1-99A9-B8BAF9897B50@freeswitch.org>

Next time we'll need to open a jira for tracking.  We have to get into  
that habit soon  ;)

/b

On Dec 26, 2008, at 11:38 AM, Adnan Barakat wrote:

> Brian West wrote:
>> Please update and try again.
>>
>> Committed revision 10949.
> Thanks Brian, works perfectly.
>
>
> Adnan
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From frank at impactfax.com  Fri Dec 26 11:57:18 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 26 Dec 2008 14:57:18 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <567c01c9677b$01739480$33014c0a@ws4>
Message-ID: <57ba01c96794$286de040$33014c0a@ws4>

I also tried this without success.  This will not fire at all regardless
of who hangs up.

<action application="set" data="api_hangup_hook=system /root/test.sh
$${base_dir}/recordings/9000_1.wav"/>
<action application="bridge" data="${enum_auto_route}"/>



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Frank @ Impact

 -----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Frank @ Impact

<action application="bridge" data="${enum_auto_route}"/>
<action application="system" data="/root/procrecording.sh
$${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
<action application="hangup"/>

The problem I am seeing is that sometimes this script gets run and
sometimes it does not.  I think it has to do maybe with which end hangs
up the phone.  But I cannot seem to nail it down just yet...







From msc at freeswitch.org  Fri Dec 26 12:06:26 2008
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Dec 2008 12:06:26 -0800
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <57ba01c96794$286de040$33014c0a@ws4>
References: <567c01c9677b$01739480$33014c0a@ws4>
	<57ba01c96794$286de040$33014c0a@ws4>
Message-ID: <87f2f3b90812261206n1be8c5e3uc91df7e34961e983@mail.gmail.com>

Frank,

I'm going to check this out as soon as I can get my test system back on
line. Thanks.
-MC

P.S. - what FS version and OS version are you on? I test with latest trunk
and CentOS 5.2

On Fri, Dec 26, 2008 at 11:57 AM, Frank @ Impact <frank at impactfax.com>wrote:

> I also tried this without success.  This will not fire at all regardless
> of who hangs up.
>
> <action application="set" data="api_hangup_hook=system /root/test.sh
> $${base_dir}/recordings/9000_1.wav"/>
> <action application="bridge" data="${enum_auto_route}"/>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Frank @ Impact
>
>  -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Frank @ Impact
>
> <action application="bridge" data="${enum_auto_route}"/>
> <action application="system" data="/root/procrecording.sh
> $${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
> <action application="hangup"/>
>
> The problem I am seeing is that sometimes this script gets run and
> sometimes it does not.  I think it has to do maybe with which end hangs
> up the phone.  But I cannot seem to nail it down just yet...
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Dec 26 12:06:37 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 15:06:37 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <567c01c9677b$01739480$33014c0a@ws4>
References: <567c01c9677b$01739480$33014c0a@ws4>
Message-ID: <CB1FDFB1-27DB-4004-8EEF-CAF82BAFDC1B@jerris.com>

This is correct, if the a leg hangs up it will not continue to run the  
dialplan actions.


On Dec 26, 2008, at 11:57 AM, Frank @ Impact wrote:

> I have confirmed that this system call does not fire if the calling
> party hangs up the phone first.  Is there a way to get the script to
> fire regardless of who hangs up first?
>
> -F
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Frank @ Impact
>
> <action application="bridge" data="${enum_auto_route}"/>
> <action application="system" data="/root/procrecording.sh
> $${base_dir}/recordings/12-26-11_10_23_55555555_444444444_1.wav"/>
> <action application="hangup"/>
>
> The problem I am seeing is that sometimes this script gets run and
> sometimes it does not.  I think it has to do maybe with which end  
> hangs
> up the phone.  But I cannot seem to nail it down just yet...




From mike at jerris.com  Fri Dec 26 12:08:17 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 15:08:17 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <57ba01c96794$286de040$33014c0a@ws4>
References: <57ba01c96794$286de040$33014c0a@ws4>
Message-ID: <572AC073-837D-4CF3-9346-F4FA49B04E91@jerris.com>

This should work, is there any debug output at hangup that would  
indicate why it doesn't run?

Mike

On Dec 26, 2008, at 2:57 PM, Frank @ Impact wrote:

> I also tried this without success.  This will not fire at all  
> regardless
> of who hangs up.
>
> <action application="set" data="api_hangup_hook=system /root/test.sh
> $${base_dir}/recordings/9000_1.wav"/>
> <action application="bridge" data="${enum_auto_route}"/>
>




From frank at impactfax.com  Fri Dec 26 12:20:45 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 26 Dec 2008 15:20:45 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <CB1FDFB1-27DB-4004-8EEF-CAF82BAFDC1B@jerris.com>
Message-ID: <57cd01c96797$6f589a10$33014c0a@ws4>

I also tried to add this

<action application="set" data="hangup_after_bridge=false"/>

to keep the dialplan process on a-leg hangup.  But that did not work
either.

Svn 10960 is what I am testing.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
 
This is correct, if the a leg hangs up it will not continue to run the  
dialplan actions.






From mike at jerris.com  Fri Dec 26 12:40:18 2008
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Dec 2008 15:40:18 -0500
Subject: [Freeswitch-users] api_hangup_hook and bash
In-Reply-To: <57cd01c96797$6f589a10$33014c0a@ws4>
References: <57cd01c96797$6f589a10$33014c0a@ws4>
Message-ID: <0BB8151C-9D6B-4F5E-BADB-DC4293A6A8B4@jerris.com>

There is no way to make the dialplan continue to run when you hang up  
the a leg, that is correct.

Mike

On Dec 26, 2008, at 3:20 PM, Frank @ Impact wrote:

> I also tried to add this
>
> <action application="set" data="hangup_after_bridge=false"/>
>
> to keep the dialplan process on a-leg hangup.  But that did not work
> either.
>
> Svn 10960 is what I am testing.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
>
> This is correct, if the a leg hangs up it will not continue to run the
> dialplan actions.
>




From juanbackson at gmail.com  Sat Dec 27 00:27:43 2008
From: juanbackson at gmail.com (Juan Backson)
Date: Sat, 27 Dec 2008 16:27:43 +0800
Subject: [Freeswitch-users] Hard limit on RTP sessions
In-Reply-To: <191c3a030812260811x56def967lc8a68755f62b7e54@mail.gmail.com>
References: <27c25bc40812260333o4067813bmdc6a63e1af4ffd6b@mail.gmail.com>
	<191c3a030812260811x56def967lc8a68755f62b7e54@mail.gmail.com>
Message-ID: <27c25bc40812270027gf6f2b8et9e1d2f9e71ac9b25@mail.gmail.com>

Hi,

Sorry for the confusion.  The 1st email was due to another problem
that I fixed, and after I fixed that problem, I am getting another
one.  Therefore, I thought the two issues are not the same.

The first issue was due to freeswich config problem related to STUN
setting.  The 2nd issue, which is what this email is about, is due to
incorrect config on sipp.  I also documented the problem and solution
in the wiki as well.

Now, I am getting another issue.  After the above two problems are
fixed, I am getting "2008-12-27 11:21:15 [ERR] switch_core_io.c:591
switch_core_session_write_frame() sofia/internal/12969 has no write
codec."  when running on a high load.

It does not happen all the time and this error does not seem to cause
call to be dropped.

Thanks for your help in advance.

JB





On Sat, Dec 27, 2008 at 12:11 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I think what's wrong is that instead of listening to the explanation on the
> other thread you already started on this issue, you started a new thread
> trying to ask the same question a different way hoping for a different
> answer.  Once you learn how to configure FreeSWITCH you will be able to stop
> using the profile with stun enabled for your testing.
>
>
> On Fri, Dec 26, 2008 at 5:33 AM, Juan Backson <juanbackson at gmail.com> wrote:
>>
>> Hi,
>>
>> Is there any hard limit set on the number of RTP sessions for
>> Freeswitch?  I am seeing freeswitch start sending out BYE after the
>> number of RTP session reaches 3000.   This problem happens even when
>> the machine utilization is still low.
>>
>> Does anyone know what is wrong?
>>
>> Thanks,
>> JB
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From yudha2008 at gmail.com  Sat Dec 27 02:56:58 2008
From: yudha2008 at gmail.com (Baskar)
Date: Sat, 27 Dec 2008 16:26:58 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
Message-ID: <d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>

*Hi **Michael,*
*
    "I try to detect the tone before answering the call.

     Is there any module for tone detect to be enabled"*

*     " I have set ignore_early_media=False **(False is case sensitive?)*"

*     But still no Tone is Detected.*

*--
Warm Regards,
N.Baskar
*
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From mike at jerris.com  Sat Dec 27 09:44:31 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 27 Dec 2008 12:44:31 -0500
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160534x715b4641u4be4a12b05213081@mail.gmail.com>
	<2ea4d47e0810160553p44ba2934peb5f9194fbd27526@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
Message-ID: <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>


On Dec 27, 2008, at 5:56 AM, Baskar wrote:

> Hi Michael,
>
>     "I try to detect the tone before answering the call.
>
>      Is there any module for tone detect to be enabled"

I believe its in mod_dptools, if its not throwing an error that the  
application does not exist, then its fine.

>
>
>      " I have set ignore_early_media=False (False is case sensitive?)"

you can omit ignore_early_media, it defaults to false.

>      But still no Tone is Detected.

Have you confirmed that tone frequency for your busy tone in your  
country and that the tone is actually being played as part of early  
media?
>
Mike

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From frank at impactfax.com  Sat Dec 27 14:21:23 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 27 Dec 2008 17:21:23 -0500
Subject: [Freeswitch-users] lua call to stop_record_session - INVALID COMMAND
Message-ID: <5e8a01c96871$73ec9f10$33014c0a@ws4>

I was trying to stop a session record from lua but when I try I get a 
"Result is INVALID COMMAND!"

I am calling this lua script with 
<action application="set" data="api_hangup_hook=lua proc.lua .."/>

so by the time the lua is called, someone has hungup one of the legs.

In the lua script I am using this to try to end the record session to
the wav file so it gets closed and so I can convert it to mp3 right away
and a few other things in the lua script...

apicmd = "stop_record_session";
apiarg = recordfile;
res = api:execute(apicmd,apiarg);

but that is when I get the INVALID COMMAND on the freeswitch console.

Is there a proper way to do this from lua?




From mike at jerris.com  Sat Dec 27 17:53:21 2008
From: mike at jerris.com (Michael Jerris)
Date: Sat, 27 Dec 2008 20:53:21 -0500
Subject: [Freeswitch-users] lua call to stop_record_session - INVALID
	COMMAND
In-Reply-To: <5e8a01c96871$73ec9f10$33014c0a@ws4>
References: <5e8a01c96871$73ec9f10$33014c0a@ws4>
Message-ID: <230AF946-21EE-4D02-912F-ABD5A734C0EB@jerris.com>


On Dec 27, 2008, at 5:21 PM, Frank @ Impact wrote:

> I was trying to stop a session record from lua but when I try I get a
> "Result is INVALID COMMAND!"
>
> I am calling this lua script with
> <action application="set" data="api_hangup_hook=lua proc.lua .."/>
>
> so by the time the lua is called, someone has hungup one of the legs.
>
> In the lua script I am using this to try to end the record session to
> the wav file so it gets closed and so I can convert it to mp3 right  
> away
> and a few other things in the lua script...
>
> apicmd = "stop_record_session";
> apiarg = recordfile;
> res = api:execute(apicmd,apiarg);
>
> but that is when I get the INVALID COMMAND on the freeswitch console.
>
> Is there a proper way to do this from lua?

When the call hangs up the record session should stop by itself  
anyways so this should not be necessary.

Mike




From wiltingtree at gmail.com  Sat Dec 27 18:15:13 2008
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sat, 27 Dec 2008 21:15:13 -0500
Subject: [Freeswitch-users] onInputCallback unstable in Python
Message-ID: <eb7797d70812271815xdf10c72v5ea1b25098205f05@mail.gmail.com>

I'm using builds 10724 and 10914 to place an outbound call from the
FreeSwitch console and use the onInputCallback functionality. My goal is to
get mod_vmd working for me.

When I run my script and press a touchtone to invoke the callback function,
I get some unstable behavior; sometimes it works fine, sometimes I get a
core dump (send me an email at wiltingtree at gmail.com if anybody wants the
core dump),  sometimes it gives me the following error:

   TypeError: onInputCallback() takes exactly 3 arguments (0 given)

In the documentation onInputCallback() takes 3 arguments, and I don't see
how it would be very useful with zero.

Here is a test script I put together which shows this behavior:

import os
from freeswitch import *
def onInputCallback(session, what, obj):
  consoleLog("INFO","IM IN THE CALLBACK!\n")
  return("continue")

def fsapi(session, stream, env, args):
  consoleLog("INFO","Hello there!!!\n")
  session =
Session("{ignore_early_media=true}sofia/gateway/gafachi/1xxxxxxxxxx")
  session.sleep(500)
  session.setInputCallback(onInputCallback)
  session.streamFile("/root/intro.wav")
  consoleLog("info","Bye!\n")
  session.hangup()
  return(session)
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From frank at impactfax.com  Sun Dec 28 08:14:27 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sun, 28 Dec 2008 11:14:27 -0500
Subject: [Freeswitch-users] lua call to stop_record_session -
	INVALIDCOMMAND
In-Reply-To: <230AF946-21EE-4D02-912F-ABD5A734C0EB@jerris.com>
Message-ID: <62c401c96907$5bad2050$33014c0a@ws4>

Mike,
I did some testing and this file is not getting closed.  I called the
script on hangup.  Made sure both legs hungup and then even did a sleep
for 5 secs to make sure FS could close any files is needed to.  Then I
made a copy of the wav file to a tmp file.  Then ended the script to
return back to the dialplan and made another copy of the wav file to a
second tmp file.

The first copy I made could not be opened by the media player. Said it
was corrupt.

The second copy of the file could be opened just fine by the media
player.

So FS is doing something to the recording file after the lua script
returns.  This is why I was trying to stop the recording session.

Any ideas?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, December 27, 2008 8:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] lua call to stop_record_session -
INVALIDCOMMAND

>
> apicmd = "stop_record_session";
> apiarg = recordfile;
> res = api:execute(apicmd,apiarg);
>
> but that is when I get the INVALID COMMAND on the freeswitch console.
>
> Is there a proper way to do this from lua?

When the call hangs up the record session should stop by itself  
anyways so this should not be necessary.

Mike






From frank at impactfax.com  Sun Dec 28 08:17:26 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sun, 28 Dec 2008 11:17:26 -0500
Subject: [Freeswitch-users] lua call to stop_record_session -
	INVALIDCOMMAND
In-Reply-To: <230AF946-21EE-4D02-912F-ABD5A734C0EB@jerris.com>
Message-ID: <62c501c96907$c648d800$33014c0a@ws4>

Another way I thought about doing this was to try to do a
api_hangup_hook=transfer stop_rec_exten.  Then in that extension, do the
stop recording and call my system processing script from there.

But I could not get the api hook to transfer to the designated extension
on hangup.  Is transfer not a valid call from api_hangup_hook?



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
 

On Dec 27, 2008, at 5:21 PM, Frank @ Impact wrote:

> I was trying to stop a session record from lua but when I try I get a
> "Result is INVALID COMMAND!"
>
When the call hangs up the record session should stop by itself  
anyways so this should not be necessary.

Mike





From frank at impactfax.com  Sun Dec 28 11:06:37 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Sun, 28 Dec 2008 14:06:37 -0500
Subject: [Freeswitch-users] session_record post-processing
Message-ID: <636b01c9691f$68802120$33014c0a@ws4>

Maybe I am going about this all wrong.  All I am trying to do is process
a recording file of a session after either one of the legs hangs up and
the call is over.  I am just trying to convert the wav to mp3 and email
it off.  So I have a bash script to do this.  The dialplan is simple
enought
 
          <action application="set" data="RECORD_ANSWER_REQ=true"/>
          <action application="set" data="RECORD_STEREO=true"/>
          <action application="record_session"
data="$${base_dir}/recordings/3329000_1.wav"/>
          <action application="bridge" data="${enum_auto_route}"/>
          <action application="hangup"/>
 
using FS svn 10960
 
But nothing I have tried seems to get it done.  I have tried to use
api_hangup_hook to call a lua script.  But the wav file is not yet
closed for some reason yet and I cannot seem to close it in the lua
script.  Basically I get the same problem if I use 'system' with the
api_hangup_hook to call the bash script to process the recording.
 
I have tried to use the transfer application with the api_hangup_hook to
allow me to stop_record_session and then a system call from another
extension, but the transfer never happens on hangup.
 
Am I missing a simple way to do this?  Is there something similar to the
'h' extension in asterisk maybe?
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From ronmccar at gmail.com  Sun Dec 28 10:52:15 2008
From: ronmccar at gmail.com (Ron McCarthy)
Date: Sun, 28 Dec 2008 11:52:15 -0700
Subject: [Freeswitch-users] Multiple context without using directory
Message-ID: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>

Hi all,

I would like to setup FS to have many context, basically we just want to
switch calls, and since we have been using Asterisk we want to keep the
context names the same, and well it's easier it seems to me.

This is for termination only, so just sending calls out, I would think FS
can do it the same way Asterisk does, where if the IP matches then it will
use those settings with that gateway and diaplan since the context ties them
all together. No matter what I try the context always goes to default, I
even took off all the ACL's for internal and external profiles and tried
gateways in each, still no luck, i can't get around the default context. We
are going to be using IP based auth only, so no user/pass's ever.

Any help on this would be great, I have searched and searched and everyone
just uses the default context, but I am trying to avoid that as the diaplan
will get quite long as I will have to all kinds of crazy matching, when a
context is much simpler, and I would think faster then using lots of
dialplan logic.

Thanks for any help on this!

Ron
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From brian at freeswitch.org  Sun Dec 28 15:20:40 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Dec 2008 17:20:40 -0600
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
Message-ID: <44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>


On Dec 28, 2008, at 12:52 PM, Ron McCarthy wrote:

> Hi all,
>
> I would like to setup FS to have many context, basically we just  
> want to switch calls, and since we have been using Asterisk we want  
> to keep the context names the same, and well it's easier it seems to  
> me.
>
> This is for termination only, so just sending calls out, I would  
> think FS can do it the same way Asterisk does, where if the IP  
> matches then it will use those settings with that gateway and  
> diaplan since the context ties them all together. No matter what I  
> try the context always goes to default, I even took off all the  
> ACL's for internal and external profiles and tried gateways in each,  
> still no luck, i can't get around the default context. We are going  
> to be using IP based auth only, so no user/pass's ever.


This type of authentication is still tied to the directory.  If you  
notice in the default config you'll notice the domains acl... that ACL  
is built off the cidr= attribute on the user tag in the directory.

   <user id="1000" cidr="1.2.3.4/24">
     <variables>
       <variable name="user_context" value="my_context"/>
     </variables>
   </user>

So now when you "reloadacl reloadxml" yes both together like that.

I also just fixed a bug related to this due to recent directory layout  
changes.

/b

PS you can also set the context to "_domain_" and it'll auto on the  
from host.


>
>
> Any help on this would be great, I have searched and searched and  
> everyone just uses the default context, but I am trying to avoid  
> that as the diaplan will get quite long as I will have to all kinds  
> of crazy matching, when a context is much simpler, and I would think  
> faster then using lots of dialplan logic.
>
> Thanks for any help on this!
>
> Ron
> _____




From Laurent.Fabre at kirranet.com  Sun Dec 28 15:36:33 2008
From: Laurent.Fabre at kirranet.com (Laurent Fabre)
Date: Mon, 29 Dec 2008 00:36:33 +0100
Subject: [Freeswitch-users] RE :  Freeswitch/Sofia configuration problem
In-Reply-To: <87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>
References: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CCC@CARBON.kirranet.msft>,
	<87f2f3b90812221522p13b770bfoc26d0242fa851e00@mail.gmail.com>
Message-ID: <DB2B7BE847E9A5408C0986C29B01CC1907AABE4CDD@CARBON.kirranet.msft>

My bad, I did RTFM and everything is working great with a lot of features turned on. I have one issue with ODBC regarding FreeTDS/SQL2005 but I googled it and it seems it's a well documented one (Invalid state cursor thingie).

If anybody managed to make things work with SQL2005 and is kind enough to share tips with me, please contact me off-list. Otherwise, I'll just fallback to some other backend solution supported by .NET.

BTW, Freeswith rocks, you guys did a tremendous work!!

Happy holidays everyone.

Regards,

-- Laurent FABRE
Directeur g?n?ral
10, rue d'Aumale
75009 Paris
Tel: +33.(0)1.42.81.28.20
Mob: +33.(0)6.75.75.02.96
Fax: +33.(0)1.70.24.74.61
laurent.fabre at kirranet.com
________________________________
De : freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] de la part de Michael Collins [msc at freeswitch.org]
Date d'envoi : mardi 23 d?cembre 2008 00:22
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] Freeswitch/Sofia configuration problem

Are you using the default config? If you've made any changes at all we'd need to know about them. Also, can you turn on SIP trace so that we can see exactly what is coming and going?
Start FS like this:
TPORT_LOG=1 ./freeswitch
Press F8 to put the console in debug mode
then capture the output while you observe the bad behavior
Please put all that, plus any config changes, into a pastebin: pastebin.freeswitch.org<http://pastebin.freeswitch.org>

I'm sure there are people around here who can help you figure out what is going on.
-MC

On Mon, Dec 22, 2008 at 8:20 AM, Laurent Fabre <Laurent.Fabre at kirranet.com<mailto:Laurent.Fabre at kirranet.com>> wrote:

Hello,

I've been trying to figure out for a few days why my freeswitch instance suddenly become insensitive to SIP packets without any warning.

What usually happen is the following :

1) start just fine in foreground mode and no errors
2) wait anywhere between 2 seconds and 20 minutes
3) Sofia suddenly decide to reload everything for some reason
4) Sofia start processing SIP packets
5) work for an hour or so
6) Sofia suddenly decide to reload everything for some reason
7) become unresponsive again
8) goto 2

Both interfaces have public IP addresses assigned in a static manner (no DHCP).

I can see the SIP UDP & TCP requests comming from the phones on several sites on the wire.
The SIP TCP requests get RST in reply which is mean :(

There was a point in my setup where it would not happen but since I'm new to freeswitch I'm having an hard time backtracking.

I was wondering if iproute/tc and iptables were the culprits but I flushed everything (even rebooted without loading the rules) and it still doesn't work.

I thought some database was corrupt so I shutdown'd freeswitch and delete his db folder, no effect.

My server runs Debian 4.0etch for amd64, built freeswitch from SVN trunk.

Any pointers, help, cure against headaches would be great :)

Regards,

Laurent


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org<mailto:Freeswitch-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ronmccar at gmail.com  Sun Dec 28 15:54:02 2008
From: ronmccar at gmail.com (Ron McCarthy)
Date: Sun, 28 Dec 2008 16:54:02 -0700
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
Message-ID: <3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>

I see the "brian.xml" example has it, didn't check that one, whoops.

Now I have added a user in the directory with the correct CIDR attribute,
yet when I send the call it seems to not use the directory, the calls is
coming from a Asterisk box, I have the Asterisk box pointed to the
"internal" profile's IP address, which I assume the directory would use, and
it gets rejected as the ACL on the incoming profile blocks that IP.

The users in the direct should not register just to IP based auth, which the
CIDR attribute takes care of?

Just looks like it's not seeing the users in the directory at all, anything
I might be missing that just jumps right out?

Thanks
Ron

On Sun, Dec 28, 2008 at 4:20 PM, Brian West <brian at freeswitch.org> wrote:

>
> On Dec 28, 2008, at 12:52 PM, Ron McCarthy wrote:
>
> > Hi all,
> >
> > I would like to setup FS to have many context, basically we just
> > want to switch calls, and since we have been using Asterisk we want
> > to keep the context names the same, and well it's easier it seems to
> > me.
> >
> > This is for termination only, so just sending calls out, I would
> > think FS can do it the same way Asterisk does, where if the IP
> > matches then it will use those settings with that gateway and
> > diaplan since the context ties them all together. No matter what I
> > try the context always goes to default, I even took off all the
> > ACL's for internal and external profiles and tried gateways in each,
> > still no luck, i can't get around the default context. We are going
> > to be using IP based auth only, so no user/pass's ever.
>
>
> This type of authentication is still tied to the directory.  If you
> notice in the default config you'll notice the domains acl... that ACL
> is built off the cidr= attribute on the user tag in the directory.
>
>   <user id="1000" cidr="1.2.3.4/24">
>     <variables>
>       <variable name="user_context" value="my_context"/>
>     </variables>
>   </user>
>
> So now when you "reloadacl reloadxml" yes both together like that.
>
> I also just fixed a bug related to this due to recent directory layout
> changes.
>
> /b
>
> PS you can also set the context to "_domain_" and it'll auto on the
> from host.
>
>
> >
> >
> > Any help on this would be great, I have searched and searched and
> > everyone just uses the default context, but I am trying to avoid
> > that as the diaplan will get quite long as I will have to all kinds
> > of crazy matching, when a context is much simpler, and I would think
> > faster then using lots of dialplan logic.
> >
> > Thanks for any help on this!
> >
> > Ron
> > _____
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Dec 28 15:58:59 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Dec 2008 17:58:59 -0600
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
Message-ID: <171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>

I can bet you're on a rev before 10981 and after 10917... update you  
have a bug related to this.

/b

On Dec 28, 2008, at 5:54 PM, Ron McCarthy wrote:

> I see the "brian.xml" example has it, didn't check that one, whoops.
>
> Now I have added a user in the directory with the correct CIDR  
> attribute, yet when I send the call it seems to not use the  
> directory, the calls is coming from a Asterisk box, I have the  
> Asterisk box pointed to the "internal" profile's IP address, which I  
> assume the directory would use, and it gets rejected as the ACL on  
> the incoming profile blocks that IP.
>
> The users in the direct should not register just to IP based auth,  
> which the CIDR attribute takes care of?
>
> Just looks like it's not seeing the users in the directory at all,  
> anything I might be missing that just jumps right out?
>
> Thanks
> Ron




From ronmccar at gmail.com  Sun Dec 28 16:16:49 2008
From: ronmccar at gmail.com (Ron McCarthy)
Date: Sun, 28 Dec 2008 17:16:49 -0700
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
	<171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
Message-ID: <3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>

Wow, 10171 I guess Im way far behind! I am building right now, ill let ya
know if any issues, thanks again!



On Sun, Dec 28, 2008 at 4:58 PM, Brian West <brian at freeswitch.org> wrote:

> I can bet you're on a rev before 10981 and after 10917... update you
> have a bug related to this.
>
> /b
>
> On Dec 28, 2008, at 5:54 PM, Ron McCarthy wrote:
>
> > I see the "brian.xml" example has it, didn't check that one, whoops.
> >
> > Now I have added a user in the directory with the correct CIDR
> > attribute, yet when I send the call it seems to not use the
> > directory, the calls is coming from a Asterisk box, I have the
> > Asterisk box pointed to the "internal" profile's IP address, which I
> > assume the directory would use, and it gets rejected as the ACL on
> > the incoming profile blocks that IP.
> >
> > The users in the direct should not register just to IP based auth,
> > which the CIDR attribute takes care of?
> >
> > Just looks like it's not seeing the users in the directory at all,
> > anything I might be missing that just jumps right out?
> >
> > Thanks
> > Ron
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Dec 28 16:28:07 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Dec 2008 18:28:07 -0600
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
	<171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
	<3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>
Message-ID: <BC41ACAF-107F-4BDD-A6B9-D3824FFDD34B@freeswitch.org>

Wow then you didn't have this one bug.... but yah a little bit  
behind ... update just to be safe.  If you still have the problem  
please post.

/b

On Dec 28, 2008, at 6:16 PM, Ron McCarthy wrote:

> Wow, 10171 I guess Im way far behind! I am building right now, ill  
> let ya know if any issues, thanks again!




From ronmccar at gmail.com  Sun Dec 28 18:46:36 2008
From: ronmccar at gmail.com (Ron McCarthy)
Date: Sun, 28 Dec 2008 19:46:36 -0700
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <BC41ACAF-107F-4BDD-A6B9-D3824FFDD34B@freeswitch.org>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
	<171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
	<3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>
	<BC41ACAF-107F-4BDD-A6B9-D3824FFDD34B@freeswitch.org>
Message-ID: <3885f4fe0812281846g74d2febcs83cc33f8aaff05f4@mail.gmail.com>

Running 10981 now, same error.

I have:

  <user id="proxy-01" cidr="x.x.xx.x/32">
    <params>
      <param name="password" value="1"/>
    </params>
    <variables>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="newContext"/>
      <variable name="effective_caller_id_name" value="CID TEST"/>
      <variable name="effective_caller_id_number" value="1000"/>
    </variables>
  </user>

Anymore ideals?

Thansk


On Sun, Dec 28, 2008 at 5:28 PM, Brian West <brian at freeswitch.org> wrote:

> Wow then you didn't have this one bug.... but yah a little bit
> behind ... update just to be safe.  If you still have the problem
> please post.
>
> /b
>
> On Dec 28, 2008, at 6:16 PM, Ron McCarthy wrote:
>
> > Wow, 10171 I guess Im way far behind! I am building right now, ill
> > let ya know if any issues, thanks again!
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Dec 28 19:02:57 2008
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Dec 2008 21:02:57 -0600
Subject: [Freeswitch-users] Multiple context without using directory
In-Reply-To: <3885f4fe0812281846g74d2febcs83cc33f8aaff05f4@mail.gmail.com>
References: <3885f4fe0812281052q4a1b51f7xa8eded5ef5151889@mail.gmail.com>
	<44668E26-5CE3-4E7F-A030-1212E9729CEB@freeswitch.org>
	<3885f4fe0812281554j1cd51efdl2ae95032f344c3e2@mail.gmail.com>
	<171B3796-2131-4EE6-ACD1-4E90A291F1F8@freeswitch.org>
	<3885f4fe0812281616s306cc7cci240422ae894cec89@mail.gmail.com>
	<BC41ACAF-107F-4BDD-A6B9-D3824FFDD34B@freeswitch.org>
	<3885f4fe0812281846g74d2febcs83cc33f8aaff05f4@mail.gmail.com>
Message-ID: <99398E29-D7F6-4ED7-91AC-0064ACA8617A@freeswitch.org>

show me the output of reloadacl  , Chances are the domain in the  
acl.conf.xml and the one in your directory don't jive.

/b

On Dec 28, 2008, at 8:46 PM, Ron McCarthy wrote:

> Running 10981 now, same error.
>
> I have:
>
>   <user id="proxy-01" cidr="x.x.xx.x/32">
>     <params>
>       <param name="password" value="1"/>
>     </params>
>     <variables>
>       <variable name="accountcode" value="1000"/>
>       <variable name="user_context" value="newContext"/>
>       <variable name="effective_caller_id_name" value="CID TEST"/>
>       <variable name="effective_caller_id_number" value="1000"/>
>     </variables>
>   </user>
>
> Anymore ideals?
>
> Thansk




From yudha2008 at gmail.com  Mon Dec 29 04:38:47 2008
From: yudha2008 at gmail.com (Baskar)
Date: Mon, 29 Dec 2008 18:08:47 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
Message-ID: <d0640d380812290438x715f5294q199f481813c76f14@mail.gmail.com>

*Hi  **Michael,*


     Step i follow for the Tone Detect process







        Thanks for the Reply It is useful for me



On Sat, Dec 27, 2008 at 11:14 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Dec 27, 2008, at 5:56 AM, Baskar wrote:
>
> *Hi **Michael,*
> *
>     "I try to detect the tone before answering the call.
>
>      Is there any module for tone detect to be enabled"*
>
>
> I believe its in mod_dptools, if its not throwing an error that the
> application does not exist, then its fine.
>
>
>
> *     " I have set ignore_early_media=False **(False is case sensitive?)*"
>
>
> you can omit ignore_early_media, it defaults to false.
>
> *     But still no Tone is Detected.*
>
>
> Have you confirmed that tone frequency for your busy tone in your country
> and that the tone is actually being played as part of early media?
>
>
>
> Mike
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Warm Regards,
N.Baskar
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From yudha2008 at gmail.com  Mon Dec 29 05:04:43 2008
From: yudha2008 at gmail.com (Baskar)
Date: Mon, 29 Dec 2008 18:34:43 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
Message-ID: <d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>

*Hi  **Michael,*

*Steps I follow for the Tone Detect process*
*
**Step1: **From X-lite i called my no (eg: 1007==>9841799874 )
**Step2: Then i run the JavaScript in that also i have given same no
(9841799874)
* *Step3: While i run the JavaScript i should get the busy tone detect but i
cant ???*

*INTERNATIONAL TELECOMMUNICATION UNION given all national frequency *

*For India they have given*

*India (Republic of)
Acceptance tone -          400   1.0 on 4.0 off
Busy tone -                    400   0.75 on 0.75 off
Congestion tone -           400   0.25 on 0.25 off
Dial tone -                      400x25   continuous
Special dial tone -           400   2.8 on 0.2 off
Holding tone -                 400   0.25 on 0.25 off 0.25 on 3.25 off
Intrusion tone -               400   0.15 on 4.85 off
Refusal tone -                400   0.25 on 0.25 off
Ringing tone - I (local calls)          400x25   0.4 on 0.2 off 0.4 on 2..0
off
Ringing tone - II (NSD/ISD calls)   400x25   1.0 on 2.0 off
Route tone -                  400   0.1 on 0.9 off
Call waiting tone -          400   0.2 on 0.1 off 0.2 on 7.5 off *


*But in wiki.sangoma They have given frequency for india (openzap)*

*[in]
generate-dial => v=-7;%(1000,0,375,425)
detect-dial => 375,425
generate-ring => v=-7;%(2000,4000,440,480)
detect-ring => 440,480
generate-busy => v=-7;%(500,500,480,620)
detect-busy => 480,620
generate-attn => v=0;%(100,100,1400,2060,2450,2600)
detect-attn => 1400,2060,2450,2600
generate-callwaiting-sas => v=0;%(300,0,440)
detect-callwaiting-sas => 440
generate-callwaiting-cas => v=0;%(80,0,2750,2130)
detect-callwaiting-cas => 2750,2130
detect-fail1 => 913.8
detect-fail2 => 1370.6
detect-fail3 => 776.7
*

*
In INTERNATIONAL TELECOMMUNICATION UNION they have given busy tone frequency
is
Busy tone -                    400   0.75 on 0.75 off

But in wiki.sangoma They have given
generate-busy => v=-7;%(500,500,480,620)
detect-busy => 480,620*

*I have tried both 400 and 480,620 in JavaScript but still i cant detect
tone.*

*Guide me Which Frequency Should i use and Another thing What steps Should
be followed for the Tone Detect .
(i)Whether my step is correct .
           OR
(ii)Should i follow in different method.If so how?  Help To detect the tone
.....*

*
Thanks for the reply,

-- 
Warm Regards,
N.Baskar
*
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From fidibus83 at aol.com  Mon Dec 29 05:20:03 2008
From: fidibus83 at aol.com (fidibus83)
Date: Mon, 29 Dec 2008 14:20:03 +0100
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
Message-ID: <00a101c969b8$29c0f420$6445310a@Franzi>

Hello,

 

I want to getting mod xml cdr working, but when I start freeswitch I get
this Error:

 

[CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error
Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so

**/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**

 

Why can?t FS load the module?

 

Thanks!

 

Best regards
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From mike at jerris.com  Mon Dec 29 05:56:46 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Dec 2008 08:56:46 -0500
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
Message-ID: <AD38E1EF-851A-4407-9C02-63207C16BB8B@jerris.com>

I don't understand your steps.

On Dec 29, 2008, at 8:04 AM, Baskar <yudha2008 at gmail.com> wrote:

> Hi  Michael,
>
> Steps I follow for the Tone Detect process
>
> Step1: From X-lite i called my no (eg: 1007==>9841799874 )
> Step2: Then i run the JavaScript in that also i have given same no (9841799874 
> )
> Step3: While i run the JavaScript i should get the busy tone detect  
> but i cant ???
>
> INTERNATIONAL TELECOMMUNICATION UNION given all national frequency
>
> For India they have given
>
> India (Republic of)
> Acceptance tone -          400   1.0 on 4.0 off
> Busy tone -                    400   0.75 on 0.75 off
> Congestion tone -           400   0.25 on 0.25 off
> Dial tone -                      400x25   continuous
> Special dial tone -           400   2.8 on 0.2 off
> Holding tone -                 400   0.25 on 0.25 off 0.25 on 3.25 off
> Intrusion tone -               400   0.15 on 4.85 off
> Refusal tone -                400   0.25 on 0.25 off
> Ringing tone - I (local calls)          400x25   0.4 on 0.2 off 0.4  
> on 2..0 off
> Ringing tone - II (NSD/ISD calls)   400x25   1.0 on 2.0 off
> Route tone -                  400   0.1 on 0.9 off
> Call waiting tone -          400   0.2 on 0.1 off 0.2 on 7.5 off
>
>
> But in wiki.sangoma They have given frequency for india (openzap)
> [in]
> generate-dial => v=-7;%(1000,0,375,425)
>
> detect-dial => 375,425
> generate-ring => v=-7;%(2000,4000,440,480)
>
> detect-ring => 440,480
> generate-busy => v=-7;%(500,500,480,620)
>
> detect-busy => 480,620
> generate-attn => v=0;%(100,100,1400,2060,2450,2600)
>
> detect-attn => 1400,2060,2450,2600
> generate-callwaiting-sas => v=0;%(300,0,440)
>
> detect-callwaiting-sas => 440
> generate-callwaiting-cas => v=0;%(80,0,2750,2130)
>
> detect-callwaiting-cas => 2750,2130
> detect-fail1 => 913.8
>
> detect-fail2 => 1370.6
> detect-fail3 => 776.7
>
> In INTERNATIONAL TELECOMMUNICATION UNION they have given busy tone  
> frequency is
> Busy tone -                    400   0.75 on 0.75 off
>
> But in wiki.sangoma They have given
> generate-busy => v=-7;%(500,500,480,620)
> detect-busy => 480,620
>
> I have tried both 400 and 480,620 in JavaScript but still i cant  
> detect tone.
>
> Guide me Which Frequency Should i use and Another thing What steps  
> Should be followed for the Tone Detect .
> (i)Whether my step is correct .
>            OR
> (ii)Should i follow in different method.If so how?  Help To detect  
> the tone .....
>
>
> Thanks for the reply,
>
> -- 
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From wiltingtree at gmail.com  Mon Dec 29 06:28:30 2008
From: wiltingtree at gmail.com (Adam Wilt)
Date: Mon, 29 Dec 2008 09:28:30 -0500
Subject: [Freeswitch-users] onInputCallback unstable in Python
In-Reply-To: <eb7797d70812271815xdf10c72v5ea1b25098205f05@mail.gmail.com>
References: <eb7797d70812271815xdf10c72v5ea1b25098205f05@mail.gmail.com>
Message-ID: <eb7797d70812290628t1eae34f4wb19313c010a595b9@mail.gmail.com>

Should I add this to Jira?



On Sat, Dec 27, 2008 at 9:15 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> I'm using builds 10724 and 10914 to place an outbound call from the
> FreeSwitch console and use the onInputCallback functionality. My goal is to
> get mod_vmd working for me.
>
> When I run my script and press a touchtone to invoke the callback
> function,  I get some unstable behavior; sometimes it works fine, sometimes
> I get a core dump (send me an email at wiltingtree at gmail.com if anybody
> wants the core dump),  sometimes it gives me the following error:
>
>    TypeError: onInputCallback() takes exactly 3 arguments (0 given)
>
> In the documentation onInputCallback() takes 3 arguments, and I don't see
> how it would be very useful with zero.
>
> Here is a test script I put together which shows this behavior:
>
> import os
> from freeswitch import *
> def onInputCallback(session, what, obj):
>   consoleLog("INFO","IM IN THE CALLBACK!\n")
>   return("continue")
>
> def fsapi(session, stream, env, args):
>   consoleLog("INFO","Hello there!!!\n")
>   session =
> Session("{ignore_early_media=true}sofia/gateway/gafachi/1xxxxxxxxxx")
>   session.sleep(500)
>   session.setInputCallback(onInputCallback)
>   session.streamFile("/root/intro.wav")
>   consoleLog("info","Bye!\n")
>   session.hangup()
>   return(session)
>
>
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From msc at freeswitch.org  Mon Dec 29 07:40:06 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 29 Dec 2008 07:40:06 -0800
Subject: [Freeswitch-users] onInputCallback unstable in Python
In-Reply-To: <eb7797d70812290628t1eae34f4wb19313c010a595b9@mail.gmail.com>
References: <eb7797d70812271815xdf10c72v5ea1b25098205f05@mail.gmail.com>
	<eb7797d70812290628t1eae34f4wb19313c010a595b9@mail.gmail.com>
Message-ID: <C84D71BF-14A3-4DFF-AA82-1E9364004473@freeswitch.org>

A jira for the core dumps would be good, especially if you can  
reproduce the behavior.

Question: where does mod_vmd come into play?

-MC

Sent from my iPhone

On Dec 29, 2008, at 6:28 AM, "Adam Wilt" <wiltingtree at gmail.com> wrote:

> Should I add this to Jira?
>
>
>
> On Sat, Dec 27, 2008 at 9:15 PM, Adam Wilt <wiltingtree at gmail.com>  
> wrote:
> I'm using builds 10724 and 10914 to place an outbound call from the  
> FreeSwitch console and use the onInputCallback functionality. My  
> goal is to get mod_vmd working for me.
>
> When I run my script and press a touchtone to invoke the callback  
> function,  I get some unstable behavior; sometimes it works fine,  
> sometimes I get a core dump (send me an email at  
> wiltingtree at gmail.com if anybody wants the core dump),  sometimes it  
> gives me the following error:
>
>    TypeError: onInputCallback() takes exactly 3 arguments (0 given)
>
> In the documentation onInputCallback() takes 3 arguments, and I  
> don't see how it would be very useful with zero.
>
> Here is a test script I put together which shows this behavior:
>
> import os
> from freeswitch import *
> def onInputCallback(session, what, obj):
>   consoleLog("INFO","IM IN THE CALLBACK!\n")
>   return("continue")
>
> def fsapi(session, stream, env, args):
>   consoleLog("INFO","Hello there!!!\n")
>   session = Session("{ignore_early_media=true}sofia/gateway/gafachi/ 
> 1xxxxxxxxxx")
>   session.sleep(500)
>   session.setInputCallback(onInputCallback)
>   session.streamFile("/root/intro.wav")
>   consoleLog("info","Bye!\n")
>   session.hangup()
>   return(session)
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From intralanman at freeswitch.org  Mon Dec 29 07:42:50 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Mon, 29 Dec 2008 10:42:50 -0500
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <20081226000150.140990@gmx.net>
References: <20081225203811.79240@gmx.net>	<2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>
	<20081226000150.140990@gmx.net>
Message-ID: <4958EFFA.2080408@freeswitch.org>

can_man at gmx.de wrote:
> Hello,
>
> thank you for your answers. I am a step further now, it seems that just the "condition" tags as described in the wiki aren't enough. After sending the following xml I think I get stuck at the point Micheal mentioned:
>
>   
>> The $$ substitutions are only done in the static XML files.  Al
>>     
>
> FS complains that: Context default not found
>
>
> XML:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="RE Dial Plan For FreeSwitch">
> <context name="public">
Pay close attention here.... notice the context name in your XML and the 
context name that FreeSWITCH is saying it can't find....
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From msc at freeswitch.org  Mon Dec 29 07:45:27 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 29 Dec 2008 07:45:27 -0800
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<191c3a030810160612t787867aepcdcdd065f864dbe8@mail.gmail.com>
	<d0640d380812180513q7ddca9ffv45e9761031feb42e@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
Message-ID: <2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>

For the sake of testing can you record a call that gets a busy signal?  
At least then we could analyze the audio and see what's going on.

If you need a dialplan example for this let me know.

-MC

Sent from my iPhone

On Dec 29, 2008, at 5:04 AM, Baskar <yudha2008 at gmail.com> wrote:

> Hi  Michael,
>
> Steps I follow for the Tone Detect process
>
> Step1: From X-lite i called my no (eg: 1007==>9841799874 )
> Step2: Then i run the JavaScript in that also i have given same no (9841799874 
> )
> Step3: While i run the JavaScript i should get the busy tone detect  
> but i cant ???
>
> INTERNATIONAL TELECOMMUNICATION UNION given all national frequency
>
> For India they have given
>
> India (Republic of)
> Acceptance tone -          400   1.0 on 4.0 off
> Busy tone -                    400   0.75 on 0.75 off
> Congestion tone -           400   0.25 on 0.25 off
> Dial tone -                      400x25   continuous
> Special dial tone -           400   2.8 on 0.2 off
> Holding tone -                 400   0.25 on 0.25 off 0.25 on 3.25 off
> Intrusion tone -               400   0.15 on 4.85 off
> Refusal tone -                400   0.25 on 0.25 off
> Ringing tone - I (local calls)          400x25   0.4 on 0.2 off 0.4  
> on 2..0 off
> Ringing tone - II (NSD/ISD calls)   400x25   1.0 on 2.0 off
> Route tone -                  400   0.1 on 0.9 off
> Call waiting tone -          400   0.2 on 0.1 off 0.2 on 7.5 off
>
>
> But in wiki.sangoma They have given frequency for india (openzap)
> [in]
> generate-dial => v=-7;%(1000,0,375,425)
>
> detect-dial => 375,425
> generate-ring => v=-7;%(2000,4000,440,480)
>
> detect-ring => 440,480
> generate-busy => v=-7;%(500,500,480,620)
>
> detect-busy => 480,620
> generate-attn => v=0;%(100,100,1400,2060,2450,2600)
>
> detect-attn => 1400,2060,2450,2600
> generate-callwaiting-sas => v=0;%(300,0,440)
>
> detect-callwaiting-sas => 440
> generate-callwaiting-cas => v=0;%(80,0,2750,2130)
>
> detect-callwaiting-cas => 2750,2130
> detect-fail1 => 913.8
>
> detect-fail2 => 1370.6
> detect-fail3 => 776.7
>
> In INTERNATIONAL TELECOMMUNICATION UNION they have given busy tone  
> frequency is
> Busy tone -                    400   0.75 on 0.75 off
>
> But in wiki.sangoma They have given
> generate-busy => v=-7;%(500,500,480,620)
> detect-busy => 480,620
>
> I have tried both 400 and 480,620 in JavaScript but still i cant  
> detect tone.
>
> Guide me Which Frequency Should i use and Another thing What steps  
> Should be followed for the Tone Detect .
> (i)Whether my step is correct .
>            OR
> (ii)Should i follow in different method.If so how?  Help To detect  
> the tone .....
>
>
> Thanks for the reply,
>
> -- 
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Mon Dec 29 07:50:42 2008
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 29 Dec 2008 07:50:42 -0800
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <636b01c9691f$68802120$33014c0a@ws4>
References: <636b01c9691f$68802120$33014c0a@ws4>
Message-ID: <1F08393E-BC33-45C3-989D-6BAD3DC40963@freeswitch.org>

I wonder if putting a sleep statement in your shell script might help.  
If it's a timing issue then possibly the shell script is trying to  
access the file before FS and/or the OS are done with it. You would  
need to tinker with how long to sleep in order to find a value that  
works in all cases.

-MC

Sent from my iPhone

On Dec 28, 2008, at 11:06 AM, "Frank @ Impact" <frank at impactfax.com>  
wrote:

> Maybe I am going about this all wrong.  All I am trying to do is  
> process a recording file of a session after either one of the legs  
> hangs up and the call is over.  I am just trying to convert the wav  
> to mp3 and email it off.  So I have a bash script to do this.  The  
> dialplan is simple enought
>
>           <action application="set" data="RECORD_ANSWER_REQ=true"/>
>           <action application="set" data="RECORD_STEREO=true"/>
>           <action application="record_session" data="$${base_dir}/ 
> recordings/3329000_1.wav"/>
>           <action application="bridge" data="${enum_auto_route}"/>
>           <action application="hangup"/>
>
>
> using FS svn 10960
>
>
>
> But nothing I have tried seems to get it done.  I have tried to use  
> api_hangup_hook to call a lua script.  But the wav file is not yet  
> closed for some reason yet and I cannot seem to close it in the lua  
> script.  Basically I get the same problem if I use ?system? with  
> the api_hangup_hook to call the bash script to process the recording.
>
>
>
> I have tried to use the transfer application with the  
> api_hangup_hook to allow me to stop_record_session and then a system  
> call from another extension, but the transfer never happens on hangup.
>
>
>
> Am I missing a simple way to do this?  Is there something similar to  
> the ?h? extension in asterisk maybe?
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From can_man at gmx.de  Mon Dec 29 08:31:52 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Mon, 29 Dec 2008 17:31:52 +0100
Subject: [Freeswitch-users] xml lib curl - transfer isn't working
In-Reply-To: <4958EFFA.2080408@freeswitch.org>
References: <20081225203811.79240@gmx.net>
	<2AE6188A-61EE-4F5E-A9C3-9FD6B0E480CD@jerris.com>
	<20081226000150.140990@gmx.net> <4958EFFA.2080408@freeswitch.org>
Message-ID: <20081229163152.206580@gmx.net>

    
> >
> > FS complains that: Context default not found
> >
> >
> > XML:
> >
> > <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> > <document type="freeswitch/xml">
> > <section name="dialplan" description="RE Dial Plan For FreeSwitch">
> > <context name="public">

> Pay close attention here.... notice the context name in your XML and the 
> context name that FreeSWITCH is saying it can't find....

>From what I understand the forward rule for my external SIP number should be in context public and the internal "music on hold" should be in context default.

However, I got everything to work now by removing the crypto checks. It works when I reply:

    '''<?xml version="1.0" encoding="UTF-8" standalone="no"?>\n'''\
    '''<document type="freeswitch/xml">\n'''\
    '''<section name="dialplan" description="RE Dial Plan For FreeSwitch">\n'''\
    '''<context name="default">\n'''\
    '''<extension name="9999">\n'''\
    '''<condition field="destination_number" expression="^(9999)$">\n'''\
    '''<action application="answer"/>\n'''\
    '''<action application="playback" data="$${hold_music}"/>\n'''\
    '''</condition>\n'''\
    '''</extension>\n'''\
    '''</context>\n'''\
    '''</section>\n'''\
    '''</document>\n''' 

But I don't understand why it works with crypto checks on when I use the xml dial plan config files and not with xml curl. Anyway, I am happy that it works now and I can continue.

Thanks for your help.
Phil
-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From mike at jerris.com  Mon Dec 29 09:10:03 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Dec 2008 12:10:03 -0500
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
In-Reply-To: <00a101c969b8$29c0f420$6445310a@Franzi>
References: <00a101c969b8$29c0f420$6445310a@Franzi>
Message-ID: <018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com>

newer fedora (core 8 and later) have libcurl that is built against  
nspr for some reason, but we don't link against it.  if you configure  
freeswitch with --without-libcurl it will use our private copy instead  
of the distro copy and resolve this issue.

Mike

On Dec 29, 2008, at 8:20 AM, fidibus83 wrote:

> Hello,
>
> I want to getting mod xml cdr working, but when I start freeswitch I  
> get this Error:
>
> [CRIT] switch_loadable_module.c:756  
> switch_loadable_module_load_file() Error Loading module /usr/local/ 
> freeswitch/mod/mod_xml_cdr.so
> **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
>
> Why can?t FS load the module?
>
> Thanks!
>
> Best regards
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frank at impactfax.com  Mon Dec 29 09:39:47 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Mon, 29 Dec 2008 12:39:47 -0500
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <1F08393E-BC33-45C3-989D-6BAD3DC40963@freeswitch.org>
Message-ID: <6a2f01c969dc$71ccf970$33014c0a@ws4>

Yes.  I had tried that.  Put a sleep 15 in the shell script before I
looked at the file.  Same results however.  FS just does not appear to
be closing that record file on hangup.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael S Collins
Sent: Monday, December 29, 2008 10:51 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session_record post-processing
 
I wonder if putting a sleep statement in your shell script might help.
If it's a timing issue then possibly the shell script is trying to
access the file before FS and/or the OS are done with it. You would need
to tinker with how long to sleep in order to find a value that works in
all cases. 
 
-MC
 
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From msc at freeswitch.org  Mon Dec 29 11:55:33 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 11:55:33 -0800
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <6a2f01c969dc$71ccf970$33014c0a@ws4>
References: <1F08393E-BC33-45C3-989D-6BAD3DC40963@freeswitch.org>
	<6a2f01c969dc$71ccf970$33014c0a@ws4>
Message-ID: <87f2f3b90812291155h3104de95if9d78344aae933f8@mail.gmail.com>

Curious: what are your endpoints? Also, what codec(s), etc. are you using?
I'm using PCMU with openzap endpoints and I don't get anything like this at
all. I'd like to try and emulate what you've got more closely to see if I
can reproduce the symptoms.

Thanks,
MC

On Mon, Dec 29, 2008 at 9:39 AM, Frank @ Impact <frank at impactfax.com> wrote:

>  Yes.  I had tried that.  Put a sleep 15 in the shell script before I
> looked at the file.  Same results however.  FS just does not appear to be
> closing that record file on hangup.
>
>
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael S
> Collins
> *Sent:* Monday, December 29, 2008 10:51 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] session_record post-processing
>
>
>
> I wonder if putting a sleep statement in your shell script might help. If
> it's a timing issue then possibly the shell script is trying to access the
> file before FS and/or the OS are done with it. You would need to tinker with
> how long to sleep in order to find a value that works in all cases.
>
>
>
> -MC
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Mon Dec 29 14:46:22 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 29 Dec 2008 23:46:22 +0100
Subject: [Freeswitch-users] event_socket and stop_dtmf
Message-ID: <4959533E.5030708@gmx.net>

When I send a stop_dtmf command via event-socket, I get a
channel_execute and a channel_execute_complete message back. However FS
still accepts DTMFs and sends them via event-socket. In addition the
other party will hear the DTMF. So I expect the stop_dtmf command is not
really executed by FS.

Here is the message I send:
SendMsg
call-command: execute
execute-app-name: stop_dtmf
execute-app-arg: true
event-lock:true
I send this command while I deliver a number of announcements to the user

At Startup I get the following on the console
2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
switch_loadable_module_process() Adding Application 'stop_dtmf'
2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
switch_loadable_module_process() Adding Application 'stop_dtmf_generate'

When I push stop_dtmf I get the following
2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event()
sofia/internal/1005 at my.domain Command Execute stop_dtmf(true)

What am I doing wrong here?

Here's a console output when I push DTMF on either side after stop_dtmf
has been pushed and the 2 call legs are bridged.
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start
packet for [1] ts=417476340 dur=160/160/2000 seq=63290
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=320/320/2000 seq=63291
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=480/480/2000 seq=63292
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=640/640/2000 seq=63293
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=800/800/2000 seq=63294
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=960/960/2000 seq=63295
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=417476340 dur=2080/2080/2000 seq=63302
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=417476340 dur=2080/2080/2000 seq=63303
2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=417476340 dur=2080/2080/2000 seq=63304
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start
packet for [1] ts=292343252 dur=160/160/2000 seq=53073
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=320/320/2000 seq=53074
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=480/480/2000 seq=53075
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=640/640/2000 seq=53076
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=800/800/2000 seq=53077
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=960/960/2000 seq=53078
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=292343252 dur=2080/2080/2000 seq=53085
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=292343252 dur=2080/2080/2000 seq=53086
2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end packet
for [1] ts=292343252 dur=2080/2080/2000 seq=53087

I am also wondering why I receive multiple events (15) for each dtmf
pressed. I expect an echo floating back and forth and triggering dtmf, hein?

Best regards
Peter




From mike at jerris.com  Mon Dec 29 15:11:23 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Dec 2008 18:11:23 -0500
Subject: [Freeswitch-users] event_socket and stop_dtmf
In-Reply-To: <4959533E.5030708@gmx.net>
References: <4959533E.5030708@gmx.net>
Message-ID: <3A19067C-6627-493D-933E-948831F8F9C1@jerris.com>

stop_dtmf is JUST for the inband dtmf listener, I would guess you are  
getting dtmf via rfc2833 or some other method.  If you want to  
understand why we generate all those packets have a read of rfc 2833.


Mike

On Dec 29, 2008, at 5:46 PM, Peter P GMX wrote:

> When I send a stop_dtmf command via event-socket, I get a
> channel_execute and a channel_execute_complete message back. However  
> FS
> still accepts DTMFs and sends them via event-socket. In addition the
> other party will hear the DTMF. So I expect the stop_dtmf command is  
> not
> really executed by FS.
>
> Here is the message I send:
> SendMsg
> call-command: execute
> execute-app-name: stop_dtmf
> execute-app-arg: true
> event-lock:true
> I send this command while I deliver a number of announcements to the  
> user
>
> At Startup I get the following on the console
> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
> switch_loadable_module_process() Adding Application 'stop_dtmf'
> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
> switch_loadable_module_process() Adding Application  
> 'stop_dtmf_generate'
>
> When I push stop_dtmf I get the following
> 2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event()
> sofia/internal/1005 at my.domain Command Execute stop_dtmf(true)
>
> What am I doing wrong here?
>
> Here's a console output when I push DTMF on either side after  
> stop_dtmf
> has been pushed and the 2 call legs are bridged.
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start
> packet for [1] ts=417476340 dur=160/160/2000 seq=63290
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=320/320/2000 seq=63291
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=480/480/2000 seq=63292
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=640/640/2000 seq=63293
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=800/800/2000 seq=63294
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=960/960/2000 seq=63295
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=417476340 dur=2080/2080/2000 seq=63302
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=417476340 dur=2080/2080/2000 seq=63303
> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=417476340 dur=2080/2080/2000 seq=63304
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start
> packet for [1] ts=292343252 dur=160/160/2000 seq=53073
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=320/320/2000 seq=53074
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=480/480/2000 seq=53075
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=640/640/2000 seq=53076
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=800/800/2000 seq=53077
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=960/960/2000 seq=53078
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
> packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=292343252 dur=2080/2080/2000 seq=53085
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=292343252 dur=2080/2080/2000 seq=53086
> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
> packet
> for [1] ts=292343252 dur=2080/2080/2000 seq=53087
>
> I am also wondering why I receive multiple events (15) for each dtmf
> pressed. I expect an echo floating back and forth and triggering  
> dtmf, hein?
>
> Best regards
> Peter
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From Prometheus001 at gmx.net  Mon Dec 29 16:41:54 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 30 Dec 2008 01:41:54 +0100
Subject: [Freeswitch-users] event_socket and stop_dtmf
In-Reply-To: <3A19067C-6627-493D-933E-948831F8F9C1@jerris.com>
References: <4959533E.5030708@gmx.net>
	<3A19067C-6627-493D-933E-948831F8F9C1@jerris.com>
Message-ID: <49596E52.1010400@gmx.net>

Yes, we get DTMF via rfc2833. If I set "dtmf-type" to "info" then
stop_dtmf works?

The reason why I want to suppress it, is the dtmf echoing (see end of my
mail). Do you see another way how I may suppress this dtmf echo? This is
severe sometimes and keeps ongoing for minutes under certain
circumstances. Thus further voice communication is no longer possible.

We have an incoming leg, play some announcements, build an outgoing leg
and then bridge those 2 channels. When any of the participiants then
pushes a dtmf, echoing begins.

Best regards
Peter

Michael Jerris schrieb:
> stop_dtmf is JUST for the inband dtmf listener, I would guess you are  
> getting dtmf via rfc2833 or some other method.  If you want to  
> understand why we generate all those packets have a read of rfc 2833.
>
>
> Mike
>
> On Dec 29, 2008, at 5:46 PM, Peter P GMX wrote:
>
>   
>> When I send a stop_dtmf command via event-socket, I get a
>> channel_execute and a channel_execute_complete message back. However  
>> FS
>> still accepts DTMFs and sends them via event-socket. In addition the
>> other party will hear the DTMF. So I expect the stop_dtmf command is  
>> not
>> really executed by FS.
>>
>> Here is the message I send:
>> SendMsg
>> call-command: execute
>> execute-app-name: stop_dtmf
>> execute-app-arg: true
>> event-lock:true
>> I send this command while I deliver a number of announcements to the  
>> user
>>
>> At Startup I get the following on the console
>> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
>> switch_loadable_module_process() Adding Application 'stop_dtmf'
>> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
>> switch_loadable_module_process() Adding Application  
>> 'stop_dtmf_generate'
>>
>> When I push stop_dtmf I get the following
>> 2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event()
>> sofia/internal/1005 at my.domain Command Execute stop_dtmf(true)
>>
>> What am I doing wrong here?
>>
>> Here's a console output when I push DTMF on either side after  
>> stop_dtmf
>> has been pushed and the 2 call legs are bridged.
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start
>> packet for [1] ts=417476340 dur=160/160/2000 seq=63290
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=320/320/2000 seq=63291
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=480/480/2000 seq=63292
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=640/640/2000 seq=63293
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=800/800/2000 seq=63294
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=960/960/2000 seq=63295
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=417476340 dur=2080/2080/2000 seq=63302
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=417476340 dur=2080/2080/2000 seq=63303
>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=417476340 dur=2080/2080/2000 seq=63304
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start
>> packet for [1] ts=292343252 dur=160/160/2000 seq=53073
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=320/320/2000 seq=53074
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=480/480/2000 seq=53075
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=640/640/2000 seq=53076
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=800/800/2000 seq=53077
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=960/960/2000 seq=53078
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>> packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=292343252 dur=2080/2080/2000 seq=53085
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=292343252 dur=2080/2080/2000 seq=53086
>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end  
>> packet
>> for [1] ts=292343252 dur=2080/2080/2000 seq=53087
>>
>> I am also wondering why I receive multiple events (15) for each dtmf
>> pressed. I expect an echo floating back and forth and triggering  
>> dtmf, hein?
>>
>> Best regards
>> Peter
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From mike at jerris.com  Mon Dec 29 17:00:20 2008
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Dec 2008 20:00:20 -0500
Subject: [Freeswitch-users] event_socket and stop_dtmf
In-Reply-To: <49596E52.1010400@gmx.net>
References: <4959533E.5030708@gmx.net>
	<3A19067C-6627-493D-933E-948831F8F9C1@jerris.com>
	<49596E52.1010400@gmx.net>
Message-ID: <0CC84B14-4BB8-48B2-8D87-A6C953624AF5@jerris.com>



On Dec 29, 2008, at 7:41 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Yes, we get DTMF via rfc2833. If I set "dtmf-type" to "info" then
> stop_dtmf works?

No, it is ONLY for inband

>
> The reason why I want to suppress it, is the dtmf echoing (see end  
> of my
> mail)

This just shows sending dtmf, not any echo.


> Do you see another way how I may suppress this dtmf echo? This is
> severe sometimes and keeps ongoing for minutes under certain
> circumstances. Thus further voice communication is no longer possible.
>

Your not getting this from dtmf in 2833.  If you have this issue you  
will probably need to fix it at the point where the dtmf is converted  
to imband.

> We have an incoming leg, play some announcements, build an outgoing  
> leg
> and then bridge those 2 channels. When any of the participiants then
> pushes a dtmf, echoing begins.
>
> Best regards
> Peter

This sounds like a Very broken provider and you should be asking them  
to correct this echo issue.

Mike
>
>
> Michael Jerris schrieb:
>> stop_dtmf is JUST for the inband dtmf listener, I would guess you are
>> getting dtmf via rfc2833 or some other method.  If you want to
>> understand why we generate all those packets have a read of rfc 2833.
>>
>>
>> Mike
>>
>> On Dec 29, 2008, at 5:46 PM, Peter P GMX wrote:
>>
>>
>>> When I send a stop_dtmf command via event-socket, I get a
>>> channel_execute and a channel_execute_complete message back. However
>>> FS
>>> still accepts DTMFs and sends them via event-socket. In addition the
>>> other party will hear the DTMF. So I expect the stop_dtmf command is
>>> not
>>> really executed by FS.
>>>
>>> Here is the message I send:
>>> SendMsg
>>> call-command: execute
>>> execute-app-name: stop_dtmf
>>> execute-app-arg: true
>>> event-lock:true
>>> I send this command while I deliver a number of announcements to the
>>> user
>>>
>>> At Startup I get the following on the console
>>> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
>>> switch_loadable_module_process() Adding Application 'stop_dtmf'
>>> 2008-12-29 22:48:05 [NOTICE] switch_loadable_module.c:238
>>> switch_loadable_module_process() Adding Application
>>> 'stop_dtmf_generate'
>>>
>>> When I push stop_dtmf I get the following
>>> 2008-12-29 22:50:10 [DEBUG] switch_ivr.c:391  
>>> switch_ivr_parse_event()
>>> sofia/internal/1005 at my.domain Command Execute stop_dtmf(true)
>>>
>>> What am I doing wrong here?
>>>
>>> Here's a console output when I push DTMF on either side after
>>> stop_dtmf
>>> has been pushed and the 2 call legs are bridged.
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1226 do_2833() Send start
>>> packet for [1] ts=417476340 dur=160/160/2000 seq=63290
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=320/320/2000 seq=63291
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=480/480/2000 seq=63292
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=640/640/2000 seq=63293
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=800/800/2000 seq=63294
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=960/960/2000 seq=63295
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1120/1120/2000 seq=63296
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1280/1280/2000 seq=63297
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1440/1440/2000 seq=63298
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1600/1600/2000 seq=63299
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1760/1760/2000 seq=63300
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=417476340 dur=1920/1920/2000 seq=63301
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=417476340 dur=2080/2080/2000 seq=63302
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=417476340 dur=2080/2080/2000 seq=63303
>>> 2008-12-29 23:13:39 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=417476340 dur=2080/2080/2000 seq=63304
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1226 do_2833() Send start
>>> packet for [1] ts=292343252 dur=160/160/2000 seq=53073
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=320/320/2000 seq=53074
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=480/480/2000 seq=53075
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=640/640/2000 seq=53076
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=800/800/2000 seq=53077
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=960/960/2000 seq=53078
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1120/1120/2000 seq=53079
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1280/1280/2000 seq=53080
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1440/1440/2000 seq=53081
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1600/1600/2000 seq=53082
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1760/1760/2000 seq=53083
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send middle
>>> packet for [1] ts=292343252 dur=1920/1920/2000 seq=53084
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=292343252 dur=2080/2080/2000 seq=53085
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=292343252 dur=2080/2080/2000 seq=53086
>>> 2008-12-29 23:13:44 [DEBUG] switch_rtp.c:1168 do_2833() Send end
>>> packet
>>> for [1] ts=292343252 dur=2080/2080/2000 seq=53087
>>>
>>> I am also wondering why I receive multiple events (15) for each dtmf
>>> pressed. I expect an echo floating back and forth and triggering
>>> dtmf, hein?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Mon Dec 29 17:19:44 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 17:19:44 -0800
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client Available!
Message-ID: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>

Hello FreeSWITCHers!

Just a heads up, there are lots of cool things happening with FreeSWITCH.
Please check out the latest here:

http://freeswitch.org/node/156

Stay tuned for more news from the FreeSWITCH camp.

-MC (mercutioviz)
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From klaus.teller at gmx.net  Mon Dec 29 17:39:38 2008
From: klaus.teller at gmx.net (Klaus Teller)
Date: Tue, 30 Dec 2008 02:39:38 +0100
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
 Available!
In-Reply-To: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
Message-ID: <20081230013938.184760@gmx.net>

This is a very much awaited tool. Thanks to you guys. Sounds like 2009 will be a very exciting year in the community.

Klaus.
-------- Original-Nachricht --------
> Datum: Mon, 29 Dec 2008 17:19:44 -0800
> Von: "Michael Collins" <msc at freeswitch.org>
> An: freeswitch-users at lists.freeswitch.org, freeswitch-dev at lists.freeswitch.org
> Betreff: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client Available!

> Hello FreeSWITCHers!
> 
> Just a heads up, there are lots of cool things happening with FreeSWITCH.
> Please check out the latest here:
> 
> http://freeswitch.org/node/156
> 
> Stay tuned for more news from the FreeSWITCH camp.
> 
> -MC (mercutioviz)

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From jason at jasonjgw.net  Mon Dec 29 18:19:42 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 30 Dec 2008 13:19:42 +1100
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH
	Client	Available!
In-Reply-To: <20081230013938.184760@gmx.net>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
	<20081230013938.184760@gmx.net>
Message-ID: <20081230021942.GA31689@jdc.jasonjgw.net>

Klaus Teller <klaus.teller at gmx.net> wrote:
> This is a very much awaited tool. Thanks to you guys. Sounds like 2009 will
> be a very exciting year in the community.

I agree.

Thanks are due to the developers for this excellent work.

I compiled it, copied fs_cli to /usr/local/bin, and now:
jason at jdc:~$ fs_cli
freeswitch at default>

Thanks!




From jason at jasonjgw.net  Mon Dec 29 20:00:49 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 30 Dec 2008 15:00:49 +1100
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH
	Client	Available!
In-Reply-To: <20081230021942.GA31689@jdc.jasonjgw.net>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
	<20081230013938.184760@gmx.net>
	<20081230021942.GA31689@jdc.jasonjgw.net>
Message-ID: <20081230040049.GA2409@jdc.jasonjgw.net>

By the way, the command to exit fs_cli is /exit (or /bye or /quit).

Commands starting with / are handled internally by the process_command()
function of the CLI, instead of being treated as FreeSWITCH API commands.




From yudha2008 at gmail.com  Mon Dec 29 22:45:06 2008
From: yudha2008 at gmail.com (Baskar)
Date: Tue, 30 Dec 2008 12:15:06 +0530
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
	<2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>
Message-ID: <d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>

Hi Michael Jerris,

I will explain what i am currently doing :I don't understand

Step 1: From the xlite phone I have dialed a number and we were on the
conversation with one extension (1007 is my extension and my mobile No
9841799874)

Step 2: From the freeswitch console I am executing a javascript file with
tone detect like the one below,

My JavaScript :

session1 = new Session();

session1.originate(session1,"{ignore_early_media=false}sofia/internal/
1003 at 172.20.201.67");

session1.execute("tone_detect","busy 480,620 r +30 transfer '1000' 3");

session1.execute("bridge", "sofia/default/9841799874 at 172.20.191.228");

session1.execute("transfer", "9841799874");

session1.hangup;
            In the above script and in step1 the telephone numbers are same;
since the script is not detecting that the phone number is busy.



Hi Michael S Collins,

            Please let me know the script, so that it would be helpful for
me.

Thanks for the Reply,

Warm Regards,
N.Baskar
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From msc at freeswitch.org  Mon Dec 29 22:53:42 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 22:53:42 -0800
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <20081230040049.GA2409@jdc.jasonjgw.net>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
	<20081230013938.184760@gmx.net>
	<20081230021942.GA31689@jdc.jasonjgw.net>
	<20081230040049.GA2409@jdc.jasonjgw.net>
Message-ID: <87f2f3b90812292253t23a67b1o784439714f0db595@mail.gmail.com>

Don't forget that there's a nice wiki page for fs_cli:
http://wiki.freeswitch.org/wiki/Fs_cli

-MC

On Dec 29, 2008, at 8:00 PM, Jason White <jason at jasonjgw.net> wrote:

> By the way, the command to exit fs_cli is /exit (or /bye or /quit).
>
> Commands starting with / are handled internally by the
> process_command()
> function of the CLI, instead of being treated as FreeSWITCH API
> commands.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From krice at suspicious.org  Mon Dec 29 22:57:45 2008
From: krice at suspicious.org (Ken Rice)
Date: Tue, 30 Dec 2008 00:57:45 -0600
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
 Available!
In-Reply-To: <20081230040049.GA2409@jdc.jasonjgw.net>
Message-ID: <C57F2289.4B7B7%krice@suspicious.org>

You can also use the ... Command to exit fs_cli and there are a few more
commands that are locally processed. (note: on the FS main console ... Will
cause fs to shutdown. Fs_cli interprets this locally and it does not shut
down the main system. You stll need to do fsclt shutdown or something
similar)

See the wiki for more information M Collins did a pretty good job
documenting it

K


> From: Jason White <jason at jasonjgw.net>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 30 Dec 2008 15:00:49 +1100
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
> Available!
> 
> By the way, the command to exit fs_cli is /exit (or /bye or /quit).
> 
> Commands starting with / are handled internally by the process_command()
> function of the CLI, instead of being treated as FreeSWITCH API commands.
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From msc at freeswitch.org  Mon Dec 29 23:21:51 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 23:21:51 -0800
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
	<2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>
	<d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>
Message-ID: <87f2f3b90812292321nfae4d0ckdc98106bf583c07@mail.gmail.com>

On Mon, Dec 29, 2008 at 10:45 PM, Baskar <yudha2008 at gmail.com> wrote:

> Hi Michael Jerris,
>
>  I will explain what i am currently doing : I don't understand
>
> Step 1: From the xlite phone I have dialed a number and we were on the
> conversation with one extension (1007 is my extension and my mobile No
> 9841799874)
>
> Step 2: From the freeswitch console I am executing a javascript file with
> tone detect like the one below,
>
> My JavaScript :
>
> session1 = new Session();
>
> session1.originate(session1,"{ignore_early_media=false}sofia/internal/
> 1003 at 172.20.201.67");
>
> session1.execute("tone_detect","busy 480,620 r +30 transfer '1000' 3");
>
I think the above line is part of the problem. The "+30" literally means
'watch for these tones for 30 milliseconds, and then don't watch any more.'
I think what you want here is +30000. Can you try that and see if there's a
difference?


> session1.execute("bridge", "sofia/default/9841799874 at 172.20.191.228");
>
> session1.execute("transfer", "9841799874");
>
> session1.hangup;
>             In the above script and in step1 the telephone numbers are
> same; since the script is not detecting that the phone number is busy.
>
>
>
> Hi Michael S Collins,
>
>             Please let me know the script, so that it would be helpful for
> me.
>
Here's a simple extension that I use for recording. It uses the uuid of the
call for the file name. I don't know if the "pre_answer" application is
absolutely necessary or not, but I do it just to be certain that I get the
early media.

<extension name="record early media">
  <condition field="destination_number" expression="^82(\d+)$"> <!-- 82 +
phone number -->
      <action application="bridge" data="sofia/default/$1 at 172.20.191.228"/>
      <!-- we can skip all this early media stuff if the far end answers -->
      <action application="set" data="execute_on_answer=transfer ANSWERCALL
XML default"/>
      <action application="sleep" data="100"/>
      <action application="pre_answer"/>
      <action application="record_session" data="/tmp/${uuid}__1.wav"/>
      <action application="sleep" data="500"/>
      <!-- If call gets here then probably we're in progress media; get
ready to handle various tones -->
      <!-- Look for busy and SIT tones right away, then wait a bit before
looking for a ring signal -->
      <action application="tone_detect" data="BUSY 480,620 r +25000 set
BUSY=true"/>  <!-- detect busy -->
      <action application="tone_detect" data="SIT1LO 913.8 r +25000 set
SIT1LO=true"/>   <!--1st segment low -->
      <!-- ... look for other SIT tones if you need to; snipped from this
example -->
      <!-- Now sleep while other end is ringing -->
      <action application="sleep" data="25010"/>
      <!-- if far end answers then we never get here...-->
      <!-- Use Lua script to check value of BUSY, SIT1LO, etc. -->
      <!-- If !BUSY and !SIT1LO, etc. then call is a no answer -->
      <!-- I use extensions for each scenario, e.g.
        <condition field="destination_number" expression="ANSWERCALL">
        OR
        <condition field="destination_number" expression="BUSYCALL">
        etc.
      -->
  </condition>
</extension>

In any case, if you call a busy number it should record the early media. You
can change the sleep time from 25010 to a shorter duration, like maybe
10000. You could also insert an info app to dump the channel variables and
see if tone_detect set your channel variable(s).

Hope this helps!
-MC

> Thanks for the Reply,
>
> Warm Regards,
> N.Baskar
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Dec 29 23:24:22 2008
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Dec 2008 23:24:22 -0800
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <C57F2289.4B7B7%krice@suspicious.org>
References: <20081230040049.GA2409@jdc.jasonjgw.net>
	<C57F2289.4B7B7%krice@suspicious.org>
Message-ID: <87f2f3b90812292324r1c070458sb20c96db5df2be53@mail.gmail.com>

Ken,

Thanks for the clarification. I will make a note of this in the wiki. Also,
can you hum a few bars and tell us what the "/filter" command does?

-MC

On Mon, Dec 29, 2008 at 10:57 PM, Ken Rice <krice at suspicious.org> wrote:

> You can also use the ... Command to exit fs_cli and there are a few more
> commands that are locally processed. (note: on the FS main console ... Will
> cause fs to shutdown. Fs_cli interprets this locally and it does not shut
> down the main system. You stll need to do fsclt shutdown or something
> similar)
>
> See the wiki for more information M Collins did a pretty good job
> documenting it
>
> K
>
>
> > From: Jason White <jason at jasonjgw.net>
> > Reply-To: <freeswitch-users at lists.freeswitch.org>
> > Date: Tue, 30 Dec 2008 15:00:49 +1100
> > To: <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
> > Available!
> >
> > By the way, the command to exit fs_cli is /exit (or /bye or /quit).
> >
> > Commands starting with / are handled internally by the process_command()
> > function of the CLI, instead of being treated as FreeSWITCH API commands.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fidibus83 at aol.com  Tue Dec 30 01:46:17 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 30 Dec 2008 10:46:17 +0100
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
In-Reply-To: <018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com>
References: <00a101c969b8$29c0f420$6445310a@Franzi>
	<018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com>
Message-ID: <008a01c96a63$76c35870$6445310a@Franzi>

Thanks for your answer. But I don?t know what I have to do now. I?m a newbie
in FS.  How do I configure FS without libcurl?

 

Thanks, fidibus

 

  _____  

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael Jerris
Gesendet: Montag, 29. Dezember 2008 18:10
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Can't load module mod_xml_cdr.so

 

newer fedora (core 8 and later) have libcurl that is built against nspr for
some reason, but we don't link against it.  if you configure freeswitch with
--without-libcurl it will use our private copy instead of the distro copy
and resolve this issue.

 

Mike

 

On Dec 29, 2008, at 8:20 AM, fidibus83 wrote:





Hello,

 

I want to getting mod xml cdr working, but when I start freeswitch I get
this Error:

 

[CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error
Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so

**/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**

 

Why can?t FS load the module?

 

Thanks!

 

Best regards

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

=
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From jason at jasonjgw.net  Tue Dec 30 01:59:50 2008
From: jason at jasonjgw.net (Jason White)
Date: Tue, 30 Dec 2008 20:59:50 +1100
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
In-Reply-To: <008a01c96a63$76c35870$6445310a@Franzi>
References: <00a101c969b8$29c0f420$6445310a@Franzi>
	<018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com>
	<008a01c96a63$76c35870$6445310a@Franzi>
Message-ID: <20081230095950.GA5884@jdc.jasonjgw.net>

fidibus83 <fidibus83 at aol.com> wrote:
> Thanks for your answer. But I don?t know what I have to do now. I?m a
> newbie in FS.  How do I configure FS without libcurl?

./configure --without-libcurl
make
(then as root)
make install

In other words, run the configure script with the --without-libcurl option,
then recompile FreeSWITCH.




From fidibus83 at aol.com  Tue Dec 30 02:38:38 2008
From: fidibus83 at aol.com (fidibus83)
Date: Tue, 30 Dec 2008 11:38:38 +0100
Subject: [Freeswitch-users] Can't load module mod_xml_cdr.so
In-Reply-To: <20081230095950.GA5884@jdc.jasonjgw.net>
References: <00a101c969b8$29c0f420$6445310a@Franzi><018FCFDC-FCF3-4883-9D1A-853DAB99582F@jerris.com><008a01c96a63$76c35870$6445310a@Franzi>
	<20081230095950.GA5884@jdc.jasonjgw.net>
Message-ID: <00b001c96a6a$c72fdc50$6445310a@Franzi>

Thanks. The Error is removed.

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jason
White
Gesendet: Dienstag, 30. Dezember 2008 11:00
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Can't load module mod_xml_cdr.so

fidibus83 <fidibus83 at aol.com> wrote:
> Thanks for your answer. But I don?t know what I have to do now. I?m a
> newbie in FS.  How do I configure FS without libcurl?

./configure --without-libcurl
make
(then as root)
make install

In other words, run the configure script with the --without-libcurl option,
then recompile FreeSWITCH.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From ivan at myrvold.org  Tue Dec 30 03:03:04 2008
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Tue, 30 Dec 2008 12:03:04 +0100
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <87f2f3b90812292324r1c070458sb20c96db5df2be53@mail.gmail.com>
References: <20081230040049.GA2409@jdc.jasonjgw.net>
	<C57F2289.4B7B7%krice@suspicious.org>
	<87f2f3b90812292324r1c070458sb20c96db5df2be53@mail.gmail.com>
Message-ID: <B95277C3-C7E9-43A7-B4F4-6B55B7BD17CE@myrvold.org>

I found out that both "/event" and "/events" worked as commands, but  
only "/noevents" worked, not "/noevent", although the Wiki says "/ 
noevent".

Ivan

Den 30. des.. 2008 kl. 08:24 skrev Michael Collins:

> Ken,
>
> Thanks for the clarification. I will make a note of this in the  
> wiki. Also, can you hum a few bars and tell us what the "/filter"  
> command does?
>
> -MC
>
> On Mon, Dec 29, 2008 at 10:57 PM, Ken Rice <krice at suspicious.org>  
> wrote:
> You can also use the ... Command to exit fs_cli and there are a few  
> more
> commands that are locally processed. (note: on the FS main  
> console ... Will
> cause fs to shutdown. Fs_cli interprets this locally and it does not  
> shut
> down the main system. You stll need to do fsclt shutdown or something
> similar)
>
> See the wiki for more information M Collins did a pretty good job
> documenting it
>
> K
>
>
> > From: Jason White <jason at jasonjgw.net>
> > Reply-To: <freeswitch-users at lists.freeswitch.org>
> > Date: Tue, 30 Dec 2008 15:00:49 +1100
> > To: <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH  
> Client
> > Available!
> >
> > By the way, the command to exit fs_cli is /exit (or /bye or /quit).
> >
> > Commands starting with / are handled internally by the  
> process_command()
> > function of the CLI, instead of being treated as FreeSWITCH API  
> commands.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kawarod at laposte.net  Tue Dec 30 04:21:32 2008
From: kawarod at laposte.net (rod)
Date: Tue, 30 Dec 2008 16:21:32 +0400
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
Message-ID: <495A124C.3040006@laposte.net>

Hi all,

I know that freeswitch has not been designed as a pure sip 
proxy/registrar, but I'm wondering how many subscribers could be handled 
by FS.

I setup the following test environment:
    - Kamailio 1.4.2 as the registrar
    - all invite requests are flowing through FS, even for a call 
between 2 registered subscribers. Many reasons for this: the calls CDR 
are centralized in the same format, I can easily add a billing ID to a 
call, proceed to recording, set the caller as anonymous if requested...
    - FS is used also as a SBC
   
There is still a lot of work to do, mainly on the call forwarding 
feature and this is why I'm wondering (simply out of curiosity) what 
could have been achieved using only FS (easier to setup when only one 
equipment is involved :) ).

I'd like to register 40 000 subscribers (if each user registers every 
60s, you have approx 670 registration per second, this setup is working 
on Kamailio).

I did the following to increase FS performance regarding registration:
    - put the directory containing users in a RAMDISK
    - put the db directory in a RAMDISK

with this I was able to reach 190 registration per second (50 without 
the ramdisk) but for one SIP account, not too useful :p (for your 
information I see a huge improvement when switching from 1.0.1 phoenix: 
150cps to FS svn 105xx: 190)
When trying with 25000 SIP accounts, I got no more than 30cps.

Then I tried to use the odbc mysql for registration, using this I was 
able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these 
tests, the presence support has been disabled.

As the IO performance seems to be a bottleneck, I'd like to know if 
there is a way to store the registration in memory only without database 
persistency.

This thread is there only to share tips, not to complain about FS poor 
performance as a SIP registrar when compared to Kamailio. If I compare 
FS to a commercial SBC I'm using in production, I have to say that FS is 
really a great piece of software (lacks only statistics module, snmp, 
and heartbeat redundancy for failover).

regards,
rod












From dyfet at gnutelephony.org  Tue Dec 30 05:07:57 2008
From: dyfet at gnutelephony.org (David Sugar)
Date: Tue, 30 Dec 2008 08:07:57 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A124C.3040006@laposte.net>
References: <495A124C.3040006@laposte.net>
Message-ID: <495A1D2D.3070507@gnutelephony.org>

You actually have potentially ~1320 effective "SIP transactions" per
second to support 40000 registered ua's with a 60s refresh.  This is
because the ua sends it's registration refresh unauthenticated.  The
registrar will then push back an authentication challenge request so the
ua can prove its identity, at which point the ua then repeats the same
transaction, but with authentication credentials attached.

rod wrote:
> Hi all,
> 
> I know that freeswitch has not been designed as a pure sip 
> proxy/registrar, but I'm wondering how many subscribers could be handled 
> by FS.
> 
> I setup the following test environment:
>     - Kamailio 1.4.2 as the registrar
>     - all invite requests are flowing through FS, even for a call 
> between 2 registered subscribers. Many reasons for this: the calls CDR 
> are centralized in the same format, I can easily add a billing ID to a 
> call, proceed to recording, set the caller as anonymous if requested...
>     - FS is used also as a SBC
>    
> There is still a lot of work to do, mainly on the call forwarding 
> feature and this is why I'm wondering (simply out of curiosity) what 
> could have been achieved using only FS (easier to setup when only one 
> equipment is involved :) ).
> 
> I'd like to register 40 000 subscribers (if each user registers every 
> 60s, you have approx 670 registration per second, this setup is working 
> on Kamailio).
> 
> I did the following to increase FS performance regarding registration:
>     - put the directory containing users in a RAMDISK
>     - put the db directory in a RAMDISK
> 
> with this I was able to reach 190 registration per second (50 without 
> the ramdisk) but for one SIP account, not too useful :p (for your 
> information I see a huge improvement when switching from 1.0.1 phoenix: 
> 150cps to FS svn 105xx: 190)
> When trying with 25000 SIP accounts, I got no more than 30cps.
> 
> Then I tried to use the odbc mysql for registration, using this I was 
> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these 
> tests, the presence support has been disabled.
> 
> As the IO performance seems to be a bottleneck, I'd like to know if 
> there is a way to store the registration in memory only without database 
> persistency.
> 
> This thread is there only to share tips, not to complain about FS poor 
> performance as a SIP registrar when compared to Kamailio. If I compare 
> FS to a commercial SBC I'm using in production, I have to say that FS is 
> really a great piece of software (lacks only statistics module, snmp, 
> and heartbeat redundancy for failover).
> 
> regards,
> rod
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gmaruzz at celliax.org  Tue Dec 30 05:27:36 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 30 Dec 2008 14:27:36 +0100
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A1D2D.3070507@gnutelephony.org>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
Message-ID: <7b197bef0812300527x5073b212j38b1a60f475440f6@mail.gmail.com>

Hi David,

very happy to read you on the FS list!

We met in 2001 at OSCon San Diego, where you "infected" me with the
telephony virus :-).

You did great work with the Bayonne project, really breaking new ground.

Thank you,

happy hacking,

happy new year!!!!



Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Tue, Dec 30, 2008 at 2:07 PM, David Sugar <dyfet at gnutelephony.org> wrote:
> You actually have potentially ~1320 effective "SIP transactions" per
> second to support 40000 registered ua's with a 60s refresh.  This is
> because the ua sends it's registration refresh unauthenticated.  The
> registrar will then push back an authentication challenge request so the
> ua can prove its identity, at which point the ua then repeats the same
> transaction, but with authentication credentials attached.
>
> rod wrote:
>> Hi all,
>>
>> I know that freeswitch has not been designed as a pure sip
>> proxy/registrar, but I'm wondering how many subscribers could be handled
>> by FS.
>>
>> I setup the following test environment:
>>     - Kamailio 1.4.2 as the registrar
>>     - all invite requests are flowing through FS, even for a call
>> between 2 registered subscribers. Many reasons for this: the calls CDR
>> are centralized in the same format, I can easily add a billing ID to a
>> call, proceed to recording, set the caller as anonymous if requested...
>>     - FS is used also as a SBC
>>
>> There is still a lot of work to do, mainly on the call forwarding
>> feature and this is why I'm wondering (simply out of curiosity) what
>> could have been achieved using only FS (easier to setup when only one
>> equipment is involved :) ).
>>
>> I'd like to register 40 000 subscribers (if each user registers every
>> 60s, you have approx 670 registration per second, this setup is working
>> on Kamailio).
>>
>> I did the following to increase FS performance regarding registration:
>>     - put the directory containing users in a RAMDISK
>>     - put the db directory in a RAMDISK
>>
>> with this I was able to reach 190 registration per second (50 without
>> the ramdisk) but for one SIP account, not too useful :p (for your
>> information I see a huge improvement when switching from 1.0.1 phoenix:
>> 150cps to FS svn 105xx: 190)
>> When trying with 25000 SIP accounts, I got no more than 30cps.
>>
>> Then I tried to use the odbc mysql for registration, using this I was
>> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these
>> tests, the presence support has been disabled.
>>
>> As the IO performance seems to be a bottleneck, I'd like to know if
>> there is a way to store the registration in memory only without database
>> persistency.
>>
>> This thread is there only to share tips, not to complain about FS poor
>> performance as a SIP registrar when compared to Kamailio. If I compare
>> FS to a commercial SBC I'm using in production, I have to say that FS is
>> really a great piece of software (lacks only statistics module, snmp,
>> and heartbeat redundancy for failover).
>>
>> regards,
>> rod
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From peder at networkoblivion.com  Tue Dec 30 05:32:16 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Tue, 30 Dec 2008 07:32:16 -0600
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A1D2D.3070507@gnutelephony.org>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
Message-ID: <495A22E0.3040904@networkoblivion.com>

>  This is
> because the ua sends it's registration refresh unauthenticated.  The
> registrar will then push back an authentication challenge request so the
> ua can prove its identity, at which point the ua then repeats the same
> transaction, but with authentication credentials attached.

Why does it do that?  Every time I do a debug, I see the first request 
denied as unauthorized and then it always comes right back and gets 
registered ok.  Is it part of the SIP spec to try unauthenticated first? 
    I would think you could set something on the UA to cut out the extra 
traffic.




From peder at networkoblivion.com  Tue Dec 30 05:37:07 2008
From: peder at networkoblivion.com (peder at networkoblivion.com)
Date: Tue, 30 Dec 2008 07:37:07 -0600
Subject: [Freeswitch-users] Register Interval
Message-ID: <495A2403.3090706@networkoblivion.com>

What do most people use as a register interval for phones?  On *, we 
always used 5 minutes and then had qualify setup, so we could keep track 
of the phones on a per minute basis as it "pinged" them every minute. 
FS doesn't do this to phones unless they are NAT'd, so if the reg is 5, 
we don't get any update for 5 minutes.  I like the idea of checking my 
phones every minute so that I know if there is a problem right away when 
someone calls with an issue, rather than having to wait up to 5 minutes 
to see if it is still alive.

Are most people using a small interval like 60 seconds?  Or do they set 
it longer and just assume the phones are still alive in between 
registrations?


Peder



From gmaruzz at celliax.org  Tue Dec 30 05:38:28 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 30 Dec 2008 14:38:28 +0100
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A22E0.3040904@networkoblivion.com>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
	<495A22E0.3040904@networkoblivion.com>
Message-ID: <7b197bef0812300538p48324e5ape8cfb1d50599e3fa@mail.gmail.com>

Yes, it is part of the SIP specs.
BTW, also HTTP works the same way.

Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Tue, Dec 30, 2008 at 2:32 PM, peder at networkoblivion.com
<peder at networkoblivion.com> wrote:
>>  This is
>> because the ua sends it's registration refresh unauthenticated.  The
>> registrar will then push back an authentication challenge request so the
>> ua can prove its identity, at which point the ua then repeats the same
>> transaction, but with authentication credentials attached.
>
> Why does it do that?  Every time I do a debug, I see the first request
> denied as unauthorized and then it always comes right back and gets
> registered ok.  Is it part of the SIP spec to try unauthenticated first?
>    I would think you could set something on the UA to cut out the extra
> traffic.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mgg at giagnocavo.net  Tue Dec 30 05:54:12 2008
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 30 Dec 2008 08:54:12 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A22E0.3040904@networkoblivion.com>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
	<495A22E0.3040904@networkoblivion.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C670233BC664C@mse17be1.mse17.exchange.ms>

>>  This is
>> because the ua sends it's registration refresh unauthenticated.  The
>> registrar will then push back an authentication challenge request so the
>> ua can prove its identity, at which point the ua then repeats the same
>> transaction, but with authentication credentials attached.
>
>Why does it do that?  Every time I do a debug, I see the first request
>denied as unauthorized and then it always comes right back and gets

Welcome to HTTP Digest authentication. The request has to get challenged to get a new nonce from the server (so as to mitigate replay attacks).

You could TLS and auth off of the client cert, except few devices support that, and you'd have the "overhead" of TCP (which is like bad or something).

-Michael



From jmesquita at gmail.com  Mon Dec 29 20:29:53 2008
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 30 Dec 2008 02:29:53 -0200
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <20081230040049.GA2409@jdc.jasonjgw.net>
References: <87f2f3b90812291719m54585dech275d231bd85f0e2d@mail.gmail.com>
	<20081230013938.184760@gmx.net>
	<20081230021942.GA31689@jdc.jasonjgw.net>
	<20081230040049.GA2409@jdc.jasonjgw.net>
Message-ID: <2A9FC6FF-41F9-47B7-8F5D-EB4FA9A0BCD7@gmail.com>

Thank you Jason, I was just going thru the code when I got your email.  
Saved me up some time. ;)

Mesquita

On Dec 30, 2008, at 2:00 AM, Jason White wrote:

> By the way, the command to exit fs_cli is /exit (or /bye or /quit).
>
> Commands starting with / are handled internally by the  
> process_command()
> function of the CLI, instead of being treated as FreeSWITCH API  
> commands.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Tue Dec 30 06:21:54 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 09:21:54 -0500
Subject: [Freeswitch-users] busy tone detection
In-Reply-To: <d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>
References: <2ea4d47e0810130602r7ed94abbqde7d85ec3ac05cb1@mail.gmail.com>
	<980C2BCA-D894-4186-BEEF-C41BFD2E2457@freeswitch.org>
	<d0640d380812240323s11bcb0e4p775295e75607d91a@mail.gmail.com>
	<87f2f3b90812240922w6445d599ucd1782209f4a4270@mail.gmail.com>
	<d0640d380812260316o3918df33seed0dd5d71211dc1@mail.gmail.com>
	<BD33F473-DF13-4093-8F7E-8D081C5CC559@jerris.com>
	<d0640d380812270256u104a99bev8b82637ba3390d58@mail.gmail.com>
	<51039E42-07E2-41B8-B684-9166BB5A4DE8@jerris.com>
	<d0640d380812290504l2c503c8ch5ee7bd7ea58d43e1@mail.gmail.com>
	<2D8229C5-9723-4235-8309-3605976EBCE6@freeswitch.org>
	<d0640d380812292245i58fe7c9aqed48c052cef0231d@mail.gmail.com>
Message-ID: <965B5A92-4182-489B-8AA0-6E6173C1931B@jerris.com>

Do you realize that calling a busy number and detecting a busy tone  
are COMPLETELY different things, your calling on sip, most likely you  
will not ever get a busy tone to detect but a sip response code when  
you try to call.

Mike

On Dec 30, 2008, at 1:45 AM, Baskar wrote:

> Hi Michael Jerris,
>
> I will explain what i am currently doing :	
> I don't understand
>
> Step 1: From the xlite phone I have dialed a number and we were on  
> the conversation with one extension (1007 is my extension and my  
> mobile No 9841799874)
> Step 2: From the freeswitch console I am executing a javascript file  
> with tone detect like the one below,
>
> My JavaScript :
>
> session1 = new Session();
>
> session1.originate(session1,"{ignore_early_media=false}sofia/internal/1003 at 172.20.201.67 
> ");
>
> session1.execute("tone_detect","busy 480,620 r +30 transfer '1000'  
> 3");
>
> session1.execute("bridge", "sofia/default/9841799874 at 172.20.191.228");
>
> session1.execute("transfer", "9841799874");
>
> session1.hangup;
>             In the above script and in step1 the telephone numbers  
> are same; since the script is not detecting that the phone number is  
> busy.
>
> Hi Michael S Collins,
>
>             Please let me know the script, so that it would be  
> helpful for me.
>
> Thanks for the Reply,
>
>
> Warm Regards,
> N.Baskar
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Dec 30 06:28:05 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 09:28:05 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A124C.3040006@laposte.net>
References: <495A124C.3040006@laposte.net>
Message-ID: <890434A9-E8B2-441F-9E88-F52B91056CA3@jerris.com>

What revision of FreeSWITCH are you trying with?  I would try with  
current trunk, I have a suspicion we fixed the main issue your running  
into.

Mike

On Dec 30, 2008, at 7:21 AM, rod wrote:

> Hi all,
>
> I know that freeswitch has not been designed as a pure sip
> proxy/registrar, but I'm wondering how many subscribers could be  
> handled
> by FS.
>
> I setup the following test environment:
>    - Kamailio 1.4.2 as the registrar
>    - all invite requests are flowing through FS, even for a call
> between 2 registered subscribers. Many reasons for this: the calls CDR
> are centralized in the same format, I can easily add a billing ID to a
> call, proceed to recording, set the caller as anonymous if  
> requested...
>    - FS is used also as a SBC
>
> There is still a lot of work to do, mainly on the call forwarding
> feature and this is why I'm wondering (simply out of curiosity) what
> could have been achieved using only FS (easier to setup when only one
> equipment is involved :) ).
>
> I'd like to register 40 000 subscribers (if each user registers every
> 60s, you have approx 670 registration per second, this setup is  
> working
> on Kamailio).
>
> I did the following to increase FS performance regarding registration:
>    - put the directory containing users in a RAMDISK
>    - put the db directory in a RAMDISK
>
> with this I was able to reach 190 registration per second (50 without
> the ramdisk) but for one SIP account, not too useful :p (for your
> information I see a huge improvement when switching from 1.0.1  
> phoenix:
> 150cps to FS svn 105xx: 190)
> When trying with 25000 SIP accounts, I got no more than 30cps.
>
> Then I tried to use the odbc mysql for registration, using this I was
> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these
> tests, the presence support has been disabled.
>
> As the IO performance seems to be a bottleneck, I'd like to know if
> there is a way to store the registration in memory only without  
> database
> persistency.
>
> This thread is there only to share tips, not to complain about FS poor
> performance as a SIP registrar when compared to Kamailio. If I compare
> FS to a commercial SBC I'm using in production, I have to say that  
> FS is
> really a great piece of software (lacks only statistics module, snmp,
> and heartbeat redundancy for failover).
>
> regards,
> rod
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From kawarod at laposte.net  Tue Dec 30 06:59:58 2008
From: kawarod at laposte.net (rod)
Date: Tue, 30 Dec 2008 18:59:58 +0400
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <890434A9-E8B2-441F-9E88-F52B91056CA3@jerris.com>
References: <495A124C.3040006@laposte.net>
	<890434A9-E8B2-441F-9E88-F52B91056CA3@jerris.com>
Message-ID: <495A376E.2060108@laposte.net>

Hi,

I upgraded today to 10999 with same results.

rod.

Michael Jerris wrote:
> What revision of FreeSWITCH are you trying with?  I would try with  
> current trunk, I have a suspicion we fixed the main issue your running  
> into.
>
> Mike
>
> On Dec 30, 2008, at 7:21 AM, rod wrote:
>
>   
>> Hi all,
>>
>> I know that freeswitch has not been designed as a pure sip
>> proxy/registrar, but I'm wondering how many subscribers could be  
>> handled
>> by FS.
>>
>> I setup the following test environment:
>>    - Kamailio 1.4.2 as the registrar
>>    - all invite requests are flowing through FS, even for a call
>> between 2 registered subscribers. Many reasons for this: the calls CDR
>> are centralized in the same format, I can easily add a billing ID to a
>> call, proceed to recording, set the caller as anonymous if  
>> requested...
>>    - FS is used also as a SBC
>>
>> There is still a lot of work to do, mainly on the call forwarding
>> feature and this is why I'm wondering (simply out of curiosity) what
>> could have been achieved using only FS (easier to setup when only one
>> equipment is involved :) ).
>>
>> I'd like to register 40 000 subscribers (if each user registers every
>> 60s, you have approx 670 registration per second, this setup is  
>> working
>> on Kamailio).
>>
>> I did the following to increase FS performance regarding registration:
>>    - put the directory containing users in a RAMDISK
>>    - put the db directory in a RAMDISK
>>
>> with this I was able to reach 190 registration per second (50 without
>> the ramdisk) but for one SIP account, not too useful :p (for your
>> information I see a huge improvement when switching from 1.0.1  
>> phoenix:
>> 150cps to FS svn 105xx: 190)
>> When trying with 25000 SIP accounts, I got no more than 30cps.
>>
>> Then I tried to use the odbc mysql for registration, using this I was
>> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these
>> tests, the presence support has been disabled.
>>
>> As the IO performance seems to be a bottleneck, I'd like to know if
>> there is a way to store the registration in memory only without  
>> database
>> persistency.
>>
>> This thread is there only to share tips, not to complain about FS poor
>> performance as a SIP registrar when compared to Kamailio. If I compare
>> FS to a commercial SBC I'm using in production, I have to say that  
>> FS is
>> really a great piece of software (lacks only statistics module, snmp,
>> and heartbeat redundancy for failover).
>>
>> regards,
>> rod
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   



From frank at impactfax.com  Tue Dec 30 07:12:28 2008
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 30 Dec 2008 10:12:28 -0500
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <87f2f3b90812291155h3104de95if9d78344aae933f8@mail.gmail.com>
Message-ID: <006c01c96a91$077d3ca0$33014c0a@ws4>

The two endpoints are sip (asterisk) and ulaw.
Thanks.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Monday, December 29, 2008 2:56 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session_record post-processing
 
Curious: what are your endpoints? Also, what codec(s), etc. are you
using? I'm using PCMU with openzap endpoints and I don't get anything
like this at all. I'd like to try and emulate what you've got more
closely to see if I can reproduce the symptoms.

Thanks,
MC
 
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From mike at jerris.com  Tue Dec 30 07:41:47 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 10:41:47 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <495A376E.2060108@laposte.net>
References: <495A124C.3040006@laposte.net>
	<890434A9-E8B2-441F-9E88-F52B91056CA3@jerris.com>
	<495A376E.2060108@laposte.net>
Message-ID: <9B4CF3F9-D10F-4270-BD2E-111D59199901@jerris.com>

If your not using sqlite, make sure to create indexes on the tables  
created, you should be able to grep the ones we do in sqlite out of  
the code.

Mike

On Dec 30, 2008, at 9:59 AM, rod wrote:

> Hi,
>
> I upgraded today to 10999 with same results.
>
> rod.
>
> Michael Jerris wrote:
>> What revision of FreeSWITCH are you trying with?  I would try with
>> current trunk, I have a suspicion we fixed the main issue your  
>> running
>> into.
>>
>> Mike
>>
>> On Dec 30, 2008, at 7:21 AM, rod wrote:
>>
>>
>>> Hi all,
>>>
>>> I know that freeswitch has not been designed as a pure sip
>>> proxy/registrar, but I'm wondering how many subscribers could be
>>> handled
>>> by FS.
>>>
>>> I setup the following test environment:
>>>   - Kamailio 1.4.2 as the registrar
>>>   - all invite requests are flowing through FS, even for a call
>>> between 2 registered subscribers. Many reasons for this: the calls  
>>> CDR
>>> are centralized in the same format, I can easily add a billing ID  
>>> to a
>>> call, proceed to recording, set the caller as anonymous if
>>> requested...
>>>   - FS is used also as a SBC
>>>
>>> There is still a lot of work to do, mainly on the call forwarding
>>> feature and this is why I'm wondering (simply out of curiosity) what
>>> could have been achieved using only FS (easier to setup when only  
>>> one
>>> equipment is involved :) ).
>>>
>>> I'd like to register 40 000 subscribers (if each user registers  
>>> every
>>> 60s, you have approx 670 registration per second, this setup is
>>> working
>>> on Kamailio).
>>>
>>> I did the following to increase FS performance regarding  
>>> registration:
>>>   - put the directory containing users in a RAMDISK
>>>   - put the db directory in a RAMDISK
>>>
>>> with this I was able to reach 190 registration per second (50  
>>> without
>>> the ramdisk) but for one SIP account, not too useful :p (for your
>>> information I see a huge improvement when switching from 1.0.1
>>> phoenix:
>>> 150cps to FS svn 105xx: 190)
>>> When trying with 25000 SIP accounts, I got no more than 30cps.
>>>
>>> Then I tried to use the odbc mysql for registration, using this I  
>>> was
>>> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all  
>>> these
>>> tests, the presence support has been disabled.
>>>
>>> As the IO performance seems to be a bottleneck, I'd like to know if
>>> there is a way to store the registration in memory only without
>>> database
>>> persistency.
>>>
>>> This thread is there only to share tips, not to complain about FS  
>>> poor
>>> performance as a SIP registrar when compared to Kamailio. If I  
>>> compare
>>> FS to a commercial SBC I'm using in production, I have to say that
>>> FS is
>>> really a great piece of software (lacks only statistics module,  
>>> snmp,
>>> and heartbeat redundancy for failover).
>>>
>>> regards,
>>> rod
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Tue Dec 30 07:46:10 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 10:46:10 -0500
Subject: [Freeswitch-users] session_record post-processing
In-Reply-To: <006c01c96a91$077d3ca0$33014c0a@ws4>
References: <006c01c96a91$077d3ca0$33014c0a@ws4>
Message-ID: <F196DCED-FCD3-410D-8FCF-94ABF2EF30FB@jerris.com>

Try svn revision 11002 or later, it should now remove the media bugs  
earlier on hangup state before the api_hangup_hook is run.

Mike

On Dec 30, 2008, at 10:12 AM, Frank @ Impact wrote:

> The two endpoints are sip (asterisk) and ulaw.
> Thanks.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Monday, December 29, 2008 2:56 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] session_record post-processing
>
> Curious: what are your endpoints? Also, what codec(s), etc. are you  
> using? I'm using PCMU with openzap endpoints and I don't get  
> anything like this at all. I'd like to try and emulate what you've  
> got more closely to see if I can reproduce the symptoms.

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From e.schmidbauer at gmail.com  Tue Dec 30 09:56:14 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 12:56:14 -0500
Subject: [Freeswitch-users] how to use celt codec
Message-ID: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>

hi, great job with the celt module. im eager to use it but im not sure how.
are there any sip clients that use the celt codec? or is there some other
way to use the celt codec to play audio in a conference? thanks.
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From brian at freeswitch.org  Tue Dec 30 10:06:59 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:06:59 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
Message-ID: <816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>

Well I used mod_portaudio on my Mac with mod_celt to my FreeSWITCH box  
on linux when we developed the module.    Works great!

/b

On Dec 30, 2008, at 11:56 AM, e schmidbauer wrote:

> hi, great job with the celt module. im eager to use it but im not  
> sure how. are there any sip clients that use the celt codec? or is  
> there some other way to use the celt codec to play audio in a  
> conference? thanks.




From can_man at gmx.de  Tue Dec 30 10:10:27 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Tue, 30 Dec 2008 19:10:27 +0100
Subject: [Freeswitch-users] voicemail - Can't find user
Message-ID: <20081230181027.267340@gmx.net>

Hello,

I am trying to get voicemail to run through xml curl, but I get the following error:

2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]

In order to setup user 315 I reply the following to the "directory" request of xml curl:

<user id="315" mailbox="315">
<params>
<param name="password" value="1234"/>
<param name="vm-password" value="0000"/>
</params>
<variables>
<variable name="accountcode" value="315"/>
<variable name="user_context" value="default"/>
<variable name="vm_extension" value="315"/>
<variable name="max_calls" value="1"/>
<variable name="fail_over" value="415"/>
<variable name="cringback" value="us-ring"/>
</variables>
</user>


And in order to send the call to voicemail I do:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="RE Dial Plan For FreeSwitch">
<context name="public">
<extension name="test10000">
<condition field="destination_number" expression="^(10000)$">
<action application="voicemail" data="default $${domain} 315"/>
</condition>
</extension>
</context>
</section>
</document>


Do I maybe have to add the user also at another location? 
Also, I read the following on the wiki: "I figured out that you can respond to both of these requests as follows. Probably the second one is looking for something different, but so far I just ignore it and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
And I do the same, I respond always with the directory response above. Is there a better practice?

It would be great if someone could point out my error. 

Thank you,
Phil


my voicemail conf looks like this:

<configuration name="voicemail.conf" description="Voicemail">
  <settings>
  </settings>
  <profiles>
    <profile name="default">
      <param name="file-extension" value="wav"/>
      <param name="terminator-key" value="#"/>
      <param name="max-login-attempts" value="3"/>
      <param name="digit-timeout" value="10000"/>
      <param name="min-record-len" value="3"/>
      <param name="max-record-len" value="300"/>
      <param name="tone-spec" value="%(1000, 0, 640)"/>
      <param name="callback-dialplan" value="XML"/>
      <param name="callback-context" value="default"/>
      <param name="play-new-messages-key" value="1"/>
      <param name="play-saved-messages-key" value="2"/>
      <param name="main-menu-key" value="0"/>
      <param name="config-menu-key" value="5"/>
      <param name="record-greeting-key" value="1"/>
      <param name="choose-greeting-key" value="2"/>
      <param name="change-pass-key" value="6"/>
      <param name="record-name-key" value="3"/>
      <param name="record-file-key" value="3"/>
      <param name="listen-file-key" value="1"/>
      <param name="save-file-key" value="2"/>
      <param name="delete-file-key" value="7"/>
      <param name="undelete-file-key" value="8"/>
      <param name="email-key" value="4"/>
      <param name="pause-key" value="0"/>
      <param name="restart-key" value="1"/>
      <param name="ff-key" value="6"/>
      <param name="rew-key" value="4"/>
      <param name="record-silence-threshold" value="200"/>
      <param name="record-silence-hits" value="2"/>
      <param name="web-template-file" value="web-vm.tpl"/>
      <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
      <!--<param name="record-sample-rate" value="11025"/>-->
      <!-- the next two both must be set for this to be enabled
           the extension is in the format of <dest> [<dialplan>] [<context>]
       -->
      <param name="operator-extension" value="operator XML default"/>
      <param name="operator-key" value="9"/>
      <param name="vmain-extension" value="vmain XML default"/>
      <param name="vmain-key" value="*"/>
      <!-- playback created files as soon as they were recorded by default -->
      <!--<param name="auto-playback-recordings" value="true"/>-->
      <email>
        <param name="template-file" value="voicemail.tpl"/>
        <param name="notify-template-file" value="notify-voicemail.tpl"/>
        <!-- this is the format voicemail_time will have -->
        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
        <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
      </email>
      <!--<param name="storage-dir" value="/tmp"/>-->
      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
      <!--<param name="record-comment" value="Your Comment"/>-->
      <!--<param name="record-title" value="Your Title"/>-->
      <!--<param name="record-copyright" value="Your Copyright"/>-->
    </profile>
  </profiles>
</configuration>






the debug output:


2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message() Asked to send early media by sofia/external/anonymous at sipgate.de
2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/anonymous at sipgate.de] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms: 20
2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms
2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message() Ring SDP:
v=0
o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
s=FreeSWITCH
c=IN IP4 89.49.116.108
t=0 0
m=audio 61125 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de!
2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de!
2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/anonymous at sipgate.de entering state [early]


2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]


2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117 switch_ivr_phrase_macro() No language specified - Using [en]
2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269 switch_ivr_phrase_macro() Handle play-file:[voicemail/vm-goodbye.wav] (en:en)
2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/external/anonymous at sipgate.de receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file
2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de [KILL]
2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE going to sleep
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_HANGUP
2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP
2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous at sipgate.de hanging up, cause: NORMAL_CLEARING
2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 480
2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP going to sleep
2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Locked, Waiting on external entities
2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Ended
2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]
-- 
Sensationsangebot verl?ngert: GMX FreeDSL - Telefonanschluss + DSL 
f?r nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a



From brian at freeswitch.org  Tue Dec 30 10:19:16 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:19:16 -0600
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <20081230181027.267340@gmx.net>
References: <20081230181027.267340@gmx.net>
Message-ID: <268644AF-1EC2-4645-9473-7E588BF25547@freeswitch.org>

what svn rev are you on?

/b

On Dec 30, 2008, at 12:10 PM, can_man at gmx.de wrote:

> <user id="315" mailbox="315">
> <params>
> <param name="password" value="1234"/>
> <param name="vm-password" value="0000"/>
> </params>
> <variables>
> <variable name="accountcode" value="315"/>
> <variable name="user_context" value="default"/>
> <variable name="vm_extension" value="315"/>
> <variable name="max_calls" value="1"/>
> <variable name="fail_over" value="415"/>
> <variable name="cringback" value="us-ring"/>
> </variables>
> </user>




From intralanman at freeswitch.org  Tue Dec 30 10:26:36 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Tue, 30 Dec 2008 18:26:36 +0000
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <20081230181027.267340@gmx.net>
References: <20081230181027.267340@gmx.net>
Message-ID: <495A67DC.9060203@freeswitch.org>

you need to add something similar to the following to your directory 
request:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="arbitrary stuff here">


-Ray




can_man at gmx.de wrote:
> Hello,
>
> I am trying to get voicemail to run through xml curl, but I get the following error:
>
> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]
>
> In order to setup user 315 I reply the following to the "directory" request of xml curl:
>
> <user id="315" mailbox="315">
> <params>
> <param name="password" value="1234"/>
> <param name="vm-password" value="0000"/>
> </params>
> <variables>
> <variable name="accountcode" value="315"/>
> <variable name="user_context" value="default"/>
> <variable name="vm_extension" value="315"/>
> <variable name="max_calls" value="1"/>
> <variable name="fail_over" value="415"/>
> <variable name="cringback" value="us-ring"/>
> </variables>
> </user>
>
>
> And in order to send the call to voicemail I do:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="RE Dial Plan For FreeSwitch">
> <context name="public">
> <extension name="test10000">
> <condition field="destination_number" expression="^(10000)$">
> <action application="voicemail" data="default $${domain} 315"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
>
> Do I maybe have to add the user also at another location? 
> Also, I read the following on the wiki: "I figured out that you can respond to both of these requests as follows. Probably the second one is looking for something different, but so far I just ignore it and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
> And I do the same, I respond always with the directory response above. Is there a better practice?
>
> It would be great if someone could point out my error. 
>
> Thank you,
> Phil
>
>
> my voicemail conf looks like this:
>
> <configuration name="voicemail.conf" description="Voicemail">
>   <settings>
>   </settings>
>   <profiles>
>     <profile name="default">
>       <param name="file-extension" value="wav"/>
>       <param name="terminator-key" value="#"/>
>       <param name="max-login-attempts" value="3"/>
>       <param name="digit-timeout" value="10000"/>
>       <param name="min-record-len" value="3"/>
>       <param name="max-record-len" value="300"/>
>       <param name="tone-spec" value="%(1000, 0, 640)"/>
>       <param name="callback-dialplan" value="XML"/>
>       <param name="callback-context" value="default"/>
>       <param name="play-new-messages-key" value="1"/>
>       <param name="play-saved-messages-key" value="2"/>
>       <param name="main-menu-key" value="0"/>
>       <param name="config-menu-key" value="5"/>
>       <param name="record-greeting-key" value="1"/>
>       <param name="choose-greeting-key" value="2"/>
>       <param name="change-pass-key" value="6"/>
>       <param name="record-name-key" value="3"/>
>       <param name="record-file-key" value="3"/>
>       <param name="listen-file-key" value="1"/>
>       <param name="save-file-key" value="2"/>
>       <param name="delete-file-key" value="7"/>
>       <param name="undelete-file-key" value="8"/>
>       <param name="email-key" value="4"/>
>       <param name="pause-key" value="0"/>
>       <param name="restart-key" value="1"/>
>       <param name="ff-key" value="6"/>
>       <param name="rew-key" value="4"/>
>       <param name="record-silence-threshold" value="200"/>
>       <param name="record-silence-hits" value="2"/>
>       <param name="web-template-file" value="web-vm.tpl"/>
>       <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
>       <!--<param name="record-sample-rate" value="11025"/>-->
>       <!-- the next two both must be set for this to be enabled
>            the extension is in the format of <dest> [<dialplan>] [<context>]
>        -->
>       <param name="operator-extension" value="operator XML default"/>
>       <param name="operator-key" value="9"/>
>       <param name="vmain-extension" value="vmain XML default"/>
>       <param name="vmain-key" value="*"/>
>       <!-- playback created files as soon as they were recorded by default -->
>       <!--<param name="auto-playback-recordings" value="true"/>-->
>       <email>
>         <param name="template-file" value="voicemail.tpl"/>
>         <param name="notify-template-file" value="notify-voicemail.tpl"/>
>         <!-- this is the format voicemail_time will have -->
>         <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>         <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
>       </email>
>       <!--<param name="storage-dir" value="/tmp"/>-->
>       <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
>       <!--<param name="record-comment" value="Your Comment"/>-->
>       <!--<param name="record-title" value="Your Title"/>-->
>       <!--<param name="record-copyright" value="Your Copyright"/>-->
>     </profile>
>   </profiles>
> </configuration>
>
>
>
>
>
>
> the debug output:
>
>
> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message() Asked to send early media by sofia/external/anonymous at sipgate.de
> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/anonymous at sipgate.de] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms: 20
> 2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms
> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message() Ring SDP:
> v=0
> o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
> s=FreeSWITCH
> c=IN IP4 89.49.116.108
> t=0 0
> m=audio 61125 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de!
> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316 sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de!
> 2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de [BREAK]
> 2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/anonymous at sipgate.de entering state [early]
>
>
> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737 voicemail_leave_main() Can't find user [315 at 192.168.178.22]
>
>
> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117 switch_ivr_phrase_macro() No language specified - Using [en]
> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269 switch_ivr_phrase_macro() Handle play-file:[voicemail/vm-goodbye.wav] (en:en)
> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
> 2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/external/anonymous at sipgate.de receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
> 2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file
> 2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de [CS_EXECUTE] [NORMAL_CLEARING]
> 2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de [KILL]
> 2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de [BREAK]
> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State EXECUTE going to sleep
> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369 switch_core_session_run() (sofia/external/anonymous at sipgate.de) Running State Change CS_HANGUP
> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP
> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/anonymous at sipgate.de hanging up, cause: NORMAL_CLEARING
> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 480
> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400 switch_core_session_run() (sofia/external/anonymous at sipgate.de) State HANGUP going to sleep
> 2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Locked, Waiting on external entities
> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de) Ended
> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de [CS_HANGUP]
>   




From brian at freeswitch.org  Tue Dec 30 10:31:45 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:31:45 -0600
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <495A67DC.9060203@freeswitch.org>
References: <20081230181027.267340@gmx.net> <495A67DC.9060203@freeswitch.org>
Message-ID: <46AB3E92-2A2C-4DF9-93F0-13D7CC314BA9@freeswitch.org>

and end that with

</section>
</document>

:P

On Dec 30, 2008, at 12:26 PM, Raymond Chandler wrote:

> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="arbitrary stuff here">




From e.schmidbauer at gmail.com  Tue Dec 30 10:33:01 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 13:33:01 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
Message-ID: <2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>

Could you explain in a more detail how you set that up?

On Tue, Dec 30, 2008 at 1:06 PM, Brian West <brian at freeswitch.org> wrote:

> Well I used mod_portaudio on my Mac with mod_celt to my FreeSWITCH box
> on linux when we developed the module.    Works great!
>
> /b
>
> On Dec 30, 2008, at 11:56 AM, e schmidbauer wrote:
>
> > hi, great job with the celt module. im eager to use it but im not
> > sure how. are there any sip clients that use the celt codec? or is
> > there some other way to use the celt codec to play audio in a
> > conference? thanks.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Dec 30 10:36:44 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:36:44 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
Message-ID: <B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>

http://wiki.freeswitch.org/wiki/Freeswitch_softphone

/b

On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:

> Could you explain in a more detail how you set that up?




From e.schmidbauer at gmail.com  Tue Dec 30 10:46:38 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 13:46:38 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
Message-ID: <2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>

i have port audio setup but when i do a 'pa call <extension>' it enters the
conference using the L16 codec. is there a way to use celt codec instead of
the L16?

On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian at freeswitch.org> wrote:

> http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>
> /b
>
> On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
>
> > Could you explain in a more detail how you set that up?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Dec 30 10:47:24 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:47:24 -0600
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <495A67DC.9060203@freeswitch.org>
References: <20081230181027.267340@gmx.net> <495A67DC.9060203@freeswitch.org>
Message-ID: <AF0F57E5-17FA-4CB5-8F76-D93E13EFBF28@freeswitch.org>

I would update to the new method using groups

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="arbitrary stuff here">
	<domain name="foo.com">
		<groups>
       		<group name="default">
         		<users>
          			<user id="315" mailbox="315">
						<params>
							<param name="password" value="1234"/>
							<param name="vm-password" value="0000"/>
						</params>
						<variables>
							<variable name="accountcode" value="315"/>
							<variable name="user_context" value="default"/>
							<variable name="vm_extension" value="315"/>
							<variable name="max_calls" value="1"/>
							<variable name="fail_over" value="415"/>
							<variable name="cringback" value="us-ring"/>
						</variables>
					</user>

         		</users>
       		</group>
		</groups>
	</domain>
</section>
</document>


/b


On Dec 30, 2008, at 12:26 PM, Raymond Chandler wrote:

> you need to add something similar to the following to your directory
> request:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="arbitrary stuff here">
>
>
> -Ray
>
>
>
>
> can_man at gmx.de wrote:
>> Hello,
>>
>> I am trying to get voicemail to run through xml curl, but I get the  
>> following error:
>>
>> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737  
>> voicemail_leave_main() Can't find user [315 at 192.168.178.22]
>>
>> In order to setup user 315 I reply the following to the "directory"  
>> request of xml curl:
>>
>> <user id="315" mailbox="315">
>> <params>
>> <param name="password" value="1234"/>
>> <param name="vm-password" value="0000"/>
>> </params>
>> <variables>
>> <variable name="accountcode" value="315"/>
>> <variable name="user_context" value="default"/>
>> <variable name="vm_extension" value="315"/>
>> <variable name="max_calls" value="1"/>
>> <variable name="fail_over" value="415"/>
>> <variable name="cringback" value="us-ring"/>
>> </variables>
>> </user>
>>
>>
>> And in order to send the call to voicemail I do:
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>> <context name="public">
>> <extension name="test10000">
>> <condition field="destination_number" expression="^(10000)$">
>> <action application="voicemail" data="default $${domain} 315"/>
>> </condition>
>> </extension>
>> </context>
>> </section>
>> </document>
>>
>>
>> Do I maybe have to add the user also at another location?
>> Also, I read the following on the wiki: "I figured out that you can  
>> respond to both of these requests as follows. Probably the second  
>> one is looking for something different, but so far I just ignore it  
>> and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
>> And I do the same, I respond always with the directory response  
>> above. Is there a better practice?
>>
>> It would be great if someone could point out my error.
>>
>> Thank you,
>> Phil
>>
>>
>> my voicemail conf looks like this:
>>
>> <configuration name="voicemail.conf" description="Voicemail">
>>  <settings>
>>  </settings>
>>  <profiles>
>>    <profile name="default">
>>      <param name="file-extension" value="wav"/>
>>      <param name="terminator-key" value="#"/>
>>      <param name="max-login-attempts" value="3"/>
>>      <param name="digit-timeout" value="10000"/>
>>      <param name="min-record-len" value="3"/>
>>      <param name="max-record-len" value="300"/>
>>      <param name="tone-spec" value="%(1000, 0, 640)"/>
>>      <param name="callback-dialplan" value="XML"/>
>>      <param name="callback-context" value="default"/>
>>      <param name="play-new-messages-key" value="1"/>
>>      <param name="play-saved-messages-key" value="2"/>
>>      <param name="main-menu-key" value="0"/>
>>      <param name="config-menu-key" value="5"/>
>>      <param name="record-greeting-key" value="1"/>
>>      <param name="choose-greeting-key" value="2"/>
>>      <param name="change-pass-key" value="6"/>
>>      <param name="record-name-key" value="3"/>
>>      <param name="record-file-key" value="3"/>
>>      <param name="listen-file-key" value="1"/>
>>      <param name="save-file-key" value="2"/>
>>      <param name="delete-file-key" value="7"/>
>>      <param name="undelete-file-key" value="8"/>
>>      <param name="email-key" value="4"/>
>>      <param name="pause-key" value="0"/>
>>      <param name="restart-key" value="1"/>
>>      <param name="ff-key" value="6"/>
>>      <param name="rew-key" value="4"/>
>>      <param name="record-silence-threshold" value="200"/>
>>      <param name="record-silence-hits" value="2"/>
>>      <param name="web-template-file" value="web-vm.tpl"/>
>>      <!-- if you need to change the sample rate of the recorded  
>> files e.g. gmail voicemail player -->
>>      <!--<param name="record-sample-rate" value="11025"/>-->
>>      <!-- the next two both must be set for this to be enabled
>>           the extension is in the format of <dest> [<dialplan>]  
>> [<context>]
>>       -->
>>      <param name="operator-extension" value="operator XML default"/>
>>      <param name="operator-key" value="9"/>
>>      <param name="vmain-extension" value="vmain XML default"/>
>>      <param name="vmain-key" value="*"/>
>>      <!-- playback created files as soon as they were recorded by  
>> default -->
>>      <!--<param name="auto-playback-recordings" value="true"/>-->
>>      <email>
>>        <param name="template-file" value="voicemail.tpl"/>
>>        <param name="notify-template-file" value="notify- 
>> voicemail.tpl"/>
>>        <!-- this is the format voicemail_time will have -->
>>        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>>        <param name="email-from" value="${voicemail_account}@$ 
>> {voicemail_domain}"/>
>>      </email>
>>      <!--<param name="storage-dir" value="/tmp"/>-->
>>      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
>>      <!--<param name="record-comment" value="Your Comment"/>-->
>>      <!--<param name="record-title" value="Your Title"/>-->
>>      <!--<param name="record-copyright" value="Your Copyright"/>-->
>>    </profile>
>>  </profiles>
>> </configuration>
>>
>>
>>
>>
>>
>>
>> the debug output:
>>
>>
>> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message()  
>> Asked to send early media by sofia/external/anonymous at sipgate.de
>> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497  
>> sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
>> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825  
>> sofia_glue_activate_rtp() AUDIO RTP [sofia/external/anonymous at sipgate.de 
>> ] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms:  
>> 20
>> 2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create()  
>> Starting timer [soft] 160 bytes per 20000ms
>> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message()  
>> Ring SDP:
>> v=0
>> o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
>> s=FreeSWITCH
>> c=IN IP4 89.49.116.108
>> t=0 0
>> m=audio 61125 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316  
>> sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de 
>> !
>> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316  
>> sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de 
>> !
>> 2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510  
>> switch_core_session_perform_receive_message() Send signal sofia/external/anonymous at sipgate.de 
>>  [BREAK]
>> 2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state()  
>> Channel sofia/external/anonymous at sipgate.de entering state [early]
>>
>>
>> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737  
>> voicemail_leave_main() Can't find user [315 at 192.168.178.22]
>>
>>
>> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117  
>> switch_ivr_phrase_macro() No language specified - Using [en]
>> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269  
>> switch_ivr_phrase_macro() Handle play-file:[voicemail/vm- 
>> goodbye.wav] (en:en)
>> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932  
>> switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
>> 2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655  
>> switch_core_session_write_frame() sofia/external/ 
>> anonymous at sipgate.de receive message  
>> [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
>> 2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222  
>> switch_ivr_play_file() done playing file
>> 2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168  
>> switch_core_standard_on_execute() Hangup sofia/external/anonymous at sipgate.de 
>>  [CS_EXECUTE] [NORMAL_CLEARING]
>> 2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494  
>> switch_channel_perform_hangup() Send signal sofia/external/anonymous at sipgate.de 
>>  [KILL]
>> 2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806  
>> switch_core_session_signal_state_change() Send signal sofia/external/anonymous at sipgate.de 
>>  [BREAK]
>> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442  
>> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
>> State EXECUTE going to sleep
>> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369  
>> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
>> Running State Change CS_HANGUP
>> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400  
>> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
>> State HANGUP
>> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup()  
>> Channel sofia/external/anonymous at sipgate.de hanging up, cause:  
>> NORMAL_CLEARING
>> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup()  
>> Responding to INVITE with: 480
>> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46  
>> switch_core_standard_on_hangup() sofia/external/ 
>> anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
>> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400  
>> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
>> State HANGUP going to sleep
>> 2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938  
>> switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de 
>> ) Locked, Waiting on external entities
>> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956  
>> switch_core_session_thread() Session 2 (sofia/external/anonymous at sipgate.de 
>> ) Ended
>> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958  
>> switch_core_session_thread() Close Channel sofia/external/anonymous at sipgate.de 
>>  [CS_HANGUP]
>>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Tue Dec 30 10:55:39 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 12:55:39 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
Message-ID: <05DFBEBE-01EE-4AA9-BC5D-B4C5403B4368@freeswitch.org>

You need to use CELT between FS and another FS box, L16 is from the PA  
to the Conference no need to encode it to celt and then decode it  
again.. it never hits the wire.

/b

On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:

> i have port audio setup but when i do a 'pa call <extension>' it  
> enters the conference using the L16 codec. is there a way to use  
> celt codec instead of the L16?




From brian at freeswitch.org  Tue Dec 30 11:06:10 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 13:06:10 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
Message-ID: <A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>

OK here try this..

in portaudio.conf.xml

     <param name="sample-rate" value="48000"/>
     <param name="codec-ms" value="10"/>


in dialplan/default.xml

     <extension name="sip_uri">
       <condition field="destination_number" expression="^sip:(.*)$">

         <action application="bridge"  
data="{absolute_codec_string=CELT at 48000h@10i}sofia/${use_profile}/sip: 
$1"/>
       </condition>
     </extension>

save that
then

pa call sip:886 at taz.bkw.org:5080

/b


On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:

> i have port audio setup but when i do a 'pa call <extension>' it  
> enters the conference using the L16 codec. is there a way to use  
> celt codec instead of the L16?
>
> On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian at freeswitch.org>  
> wrote:
> http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>
> /b
>
> On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
>
> > Could you explain in a more detail how you set that up?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From can_man at gmx.de  Tue Dec 30 11:08:01 2008
From: can_man at gmx.de (can_man at gmx.de)
Date: Tue, 30 Dec 2008 20:08:01 +0100
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <AF0F57E5-17FA-4CB5-8F76-D93E13EFBF28@freeswitch.org>
References: <20081230181027.267340@gmx.net> <495A67DC.9060203@freeswitch.org>
	<AF0F57E5-17FA-4CB5-8F76-D93E13EFBF28@freeswitch.org>
Message-ID: <20081230190801.302740@gmx.net>

Hello,

thank you for your answers. I have added the start and end tags to my xml, but nothing has changed.
However, the "groups XML" did work with my server's IP as: <domain name="192.168.178.22"> - thank you Brian.

If someone can shade some light into this quote from the wiki:  "I figured out that you can respond to both of these requests as follows. Probably the second one is looking for something different, but so far I just ignore it and throw out the same stuff." at http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22

I will re-write the whole "directory" section on the wiki. For now I will add the "group" reply.

Thank you,
Phil

Ps: if it is still of interest, my svn version is:

URL: http://svn.freeswitch.org/svn/freeswitch/trunk
Repository Root: http://svn.freeswitch.org/svn
Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2
Revision: 10988
Node Kind: directory
Schedule: normal
Last Changed Author: brian
Last Changed Rev: 10983
Last Changed Date: 2008-12-29 06:27:53 +0100 (Mon, 29 Dec 2008)




> I would update to the new method using groups
> 
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="arbitrary stuff here">
> 	<domain name="foo.com">
> 		<groups>
>        		<group name="default">
>          		<users>
>           			<user id="315" mailbox="315">
> 						<params>
> 							<param name="password" value="1234"/>
> 							<param name="vm-password" value="0000"/>
> 						</params>
> 						<variables>
> 							<variable name="accountcode" value="315"/>
> 							<variable name="user_context" value="default"/>
> 							<variable name="vm_extension" value="315"/>
> 							<variable name="max_calls" value="1"/>
> 							<variable name="fail_over" value="415"/>
> 							<variable name="cringback" value="us-ring"/>
> 						</variables>
> 					</user>
> 
>          		</users>
>        		</group>
> 		</groups>
> 	</domain>
> </section>
> </document>
> 
> 
> /b
> 
> 
> On Dec 30, 2008, at 12:26 PM, Raymond Chandler wrote:
> 
> > you need to add something similar to the following to your directory
> > request:
> >
> > <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> > <document type="freeswitch/xml">
> > <section name="directory" description="arbitrary stuff here">
> >
> >
> > -Ray
> >
> >
> >
> >
> > can_man at gmx.de wrote:
> >> Hello,
> >>
> >> I am trying to get voicemail to run through xml curl, but I get the  
> >> following error:
> >>
> >> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737  
> >> voicemail_leave_main() Can't find user [315 at 192.168.178.22]
> >>
> >> In order to setup user 315 I reply the following to the "directory"  
> >> request of xml curl:
> >>
> >> <user id="315" mailbox="315">
> >> <params>
> >> <param name="password" value="1234"/>
> >> <param name="vm-password" value="0000"/>
> >> </params>
> >> <variables>
> >> <variable name="accountcode" value="315"/>
> >> <variable name="user_context" value="default"/>
> >> <variable name="vm_extension" value="315"/>
> >> <variable name="max_calls" value="1"/>
> >> <variable name="fail_over" value="415"/>
> >> <variable name="cringback" value="us-ring"/>
> >> </variables>
> >> </user>
> >>
> >>
> >> And in order to send the call to voicemail I do:
> >>
> >> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> >> <document type="freeswitch/xml">
> >> <section name="dialplan" description="RE Dial Plan For FreeSwitch">
> >> <context name="public">
> >> <extension name="test10000">
> >> <condition field="destination_number" expression="^(10000)$">
> >> <action application="voicemail" data="default $${domain} 315"/>
> >> </condition>
> >> </extension>
> >> </context>
> >> </section>
> >> </document>
> >>
> >>
> >> Do I maybe have to add the user also at another location?
> >> Also, I read the following on the wiki: "I figured out that you can  
> >> respond to both of these requests as follows. Probably the second  
> >> one is looking for something different, but so far I just ignore it  
> >> and throw out the same stuff." at
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
> >> And I do the same, I respond always with the directory response  
> >> above. Is there a better practice?
> >>
> >> It would be great if someone could point out my error.
> >>
> >> Thank you,
> >> Phil
> >>
> >>
> >> my voicemail conf looks like this:
> >>
> >> <configuration name="voicemail.conf" description="Voicemail">
> >>  <settings>
> >>  </settings>
> >>  <profiles>
> >>    <profile name="default">
> >>      <param name="file-extension" value="wav"/>
> >>      <param name="terminator-key" value="#"/>
> >>      <param name="max-login-attempts" value="3"/>
> >>      <param name="digit-timeout" value="10000"/>
> >>      <param name="min-record-len" value="3"/>
> >>      <param name="max-record-len" value="300"/>
> >>      <param name="tone-spec" value="%(1000, 0, 640)"/>
> >>      <param name="callback-dialplan" value="XML"/>
> >>      <param name="callback-context" value="default"/>
> >>      <param name="play-new-messages-key" value="1"/>
> >>      <param name="play-saved-messages-key" value="2"/>
> >>      <param name="main-menu-key" value="0"/>
> >>      <param name="config-menu-key" value="5"/>
> >>      <param name="record-greeting-key" value="1"/>
> >>      <param name="choose-greeting-key" value="2"/>
> >>      <param name="change-pass-key" value="6"/>
> >>      <param name="record-name-key" value="3"/>
> >>      <param name="record-file-key" value="3"/>
> >>      <param name="listen-file-key" value="1"/>
> >>      <param name="save-file-key" value="2"/>
> >>      <param name="delete-file-key" value="7"/>
> >>      <param name="undelete-file-key" value="8"/>
> >>      <param name="email-key" value="4"/>
> >>      <param name="pause-key" value="0"/>
> >>      <param name="restart-key" value="1"/>
> >>      <param name="ff-key" value="6"/>
> >>      <param name="rew-key" value="4"/>
> >>      <param name="record-silence-threshold" value="200"/>
> >>      <param name="record-silence-hits" value="2"/>
> >>      <param name="web-template-file" value="web-vm.tpl"/>
> >>      <!-- if you need to change the sample rate of the recorded  
> >> files e.g. gmail voicemail player -->
> >>      <!--<param name="record-sample-rate" value="11025"/>-->
> >>      <!-- the next two both must be set for this to be enabled
> >>           the extension is in the format of <dest> [<dialplan>]  
> >> [<context>]
> >>       -->
> >>      <param name="operator-extension" value="operator XML default"/>
> >>      <param name="operator-key" value="9"/>
> >>      <param name="vmain-extension" value="vmain XML default"/>
> >>      <param name="vmain-key" value="*"/>
> >>      <!-- playback created files as soon as they were recorded by  
> >> default -->
> >>      <!--<param name="auto-playback-recordings" value="true"/>-->
> >>      <email>
> >>        <param name="template-file" value="voicemail.tpl"/>
> >>        <param name="notify-template-file" value="notify- 
> >> voicemail.tpl"/>
> >>        <!-- this is the format voicemail_time will have -->
> >>        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
> >>        <param name="email-from" value="${voicemail_account}@$ 
> >> {voicemail_domain}"/>
> >>      </email>
> >>      <!--<param name="storage-dir" value="/tmp"/>-->
> >>      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
> >>      <!--<param name="record-comment" value="Your Comment"/>-->
> >>      <!--<param name="record-title" value="Your Title"/>-->
> >>      <!--<param name="record-copyright" value="Your Copyright"/>-->
> >>    </profile>
> >>  </profiles>
> >> </configuration>
> >>
> >>
> >>
> >>
> >>
> >>
> >> the debug output:
> >>
> >>
> >> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1272 sofia_receive_message()  
> >> Asked to send early media by sofia/external/anonymous at sipgate.de
> >> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:497  
> >> sofia_glue_ext_address_lookup() STUN Success [89.49.116.108]:[61125]
> >> 2008-12-30 18:41:54 [DEBUG] sofia_glue.c:1825  
> >> sofia_glue_activate_rtp() AUDIO RTP
> [sofia/external/anonymous at sipgate.de 
> >> ] 192.168.178.22 port 25060 -> 217.10.77.21 port 57708 codec: 8 ms:  
> >> 20
> >> 2008-12-30 18:41:54 [DEBUG] switch_rtp.c:859 switch_rtp_create()  
> >> Starting timer [soft] 160 bytes per 20000ms
> >> 2008-12-30 18:41:54 [INFO] mod_sofia.c:1313 sofia_receive_message()  
> >> Ring SDP:
> >> v=0
> >> o=FreeSWITCH 1230597789 1230597790 IN IP4 89.49.116.108
> >> s=FreeSWITCH
> >> c=IN IP4 89.49.116.108
> >> t=0 0
> >> m=audio 61125 RTP/AVP 8 101
> >> a=rtpmap:8 PCMA/8000
> >> a=rtpmap:101 telephone-event/8000
> >> a=fmtp:101 0-16
> >> a=silenceSupp:off - - - -
> >> a=ptime:20
> >> a=sendrecv
> >>
> >> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316  
> >> sofia_receive_message() Ring-Ready sofia/external/anonymous at sipgate.de 
> >> !
> >> 2008-12-30 18:41:54 [NOTICE] mod_sofia.c:1316  
> >> sofia_receive_message() Pre-Answer sofia/external/anonymous at sipgate.de 
> >> !
> >> 2008-12-30 18:41:54 [DEBUG] switch_core_session.c:510  
> >> switch_core_session_perform_receive_message() Send signal
> sofia/external/anonymous at sipgate.de 
> >>  [BREAK]
> >> 2008-12-30 18:41:54 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state()  
> >> Channel sofia/external/anonymous at sipgate.de entering state [early]
> >>
> >>
> >> 2008-12-30 18:41:54 [WARNING] mod_voicemail.c:2737  
> >> voicemail_leave_main() Can't find user [315 at 192.168.178.22]
> >>
> >>
> >> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:117  
> >> switch_ivr_phrase_macro() No language specified - Using [en]
> >> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:269  
> >> switch_ivr_phrase_macro() Handle play-file:[voicemail/vm- 
> >> goodbye.wav] (en:en)
> >> 2008-12-30 18:41:54 [DEBUG] switch_ivr_play_say.c:932  
> >> switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
> >> 2008-12-30 18:41:54 [DEBUG] switch_core_io.c:655  
> >> switch_core_session_write_frame() sofia/external/ 
> >> anonymous at sipgate.de receive message  
> >> [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY]
> >> 2008-12-30 18:41:55 [DEBUG] switch_ivr_play_say.c:1222  
> >> switch_ivr_play_file() done playing file
> >> 2008-12-30 18:41:55 [NOTICE] switch_core_state_machine.c:168  
> >> switch_core_standard_on_execute() Hangup
> sofia/external/anonymous at sipgate.de 
> >>  [CS_EXECUTE] [NORMAL_CLEARING]
> >> 2008-12-30 18:41:55 [DEBUG] switch_channel.c:1494  
> >> switch_channel_perform_hangup() Send signal
> sofia/external/anonymous at sipgate.de 
> >>  [KILL]
> >> 2008-12-30 18:41:55 [DEBUG] switch_core_session.c:806  
> >> switch_core_session_signal_state_change() Send signal
> sofia/external/anonymous at sipgate.de 
> >>  [BREAK]
> >> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:442  
> >> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
> >> State EXECUTE going to sleep
> >> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:369  
> >> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
> >> Running State Change CS_HANGUP
> >> 2008-12-30 18:41:55 [DEBUG] switch_core_state_machine.c:400  
> >> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
> >> State HANGUP
> >> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:287 sofia_on_hangup()  
> >> Channel sofia/external/anonymous at sipgate.de hanging up, cause:  
> >> NORMAL_CLEARING
> >> 2008-12-30 18:41:55 [DEBUG] mod_sofia.c:361 sofia_on_hangup()  
> >> Responding to INVITE with: 480
> >> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:46  
> >> switch_core_standard_on_hangup() sofia/external/ 
> >> anonymous at sipgate.de Standard HANGUP, cause: NORMAL_CLEARING
> >> 2008-12-30 18:41:57 [DEBUG] switch_core_state_machine.c:400  
> >> switch_core_session_run() (sofia/external/anonymous at sipgate.de)  
> >> State HANGUP going to sleep
> >> 2008-12-30 18:41:57 [DEBUG] switch_core_session.c:938  
> >> switch_core_session_thread() Session 2
> (sofia/external/anonymous at sipgate.de 
> >> ) Locked, Waiting on external entities
> >> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:956  
> >> switch_core_session_thread() Session 2
> (sofia/external/anonymous at sipgate.de 
> >> ) Ended
> >> 2008-12-30 18:41:57 [NOTICE] switch_core_session.c:958  
> >> switch_core_session_thread() Close Channel
> sofia/external/anonymous at sipgate.de 
> >>  [CS_HANGUP]
> >>
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger



From brian at freeswitch.org  Tue Dec 30 11:14:23 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 13:14:23 -0600
Subject: [Freeswitch-users] voicemail - Can't find user
In-Reply-To: <20081230190801.302740@gmx.net>
References: <20081230181027.267340@gmx.net> <495A67DC.9060203@freeswitch.org>
	<AF0F57E5-17FA-4CB5-8F76-D93E13EFBF28@freeswitch.org>
	<20081230190801.302740@gmx.net>
Message-ID: <91A83654-30D9-4DDC-A93F-E9B3109BEF65@freeswitch.org>

Thank you... it needed to be updated ;)

btw we now have a group/ endpoint.. so you can call group/sales at domain  
and ring everyone in the group.  Go check out the default config it  
has use examples.

/b

On Dec 30, 2008, at 1:08 PM, can_man at gmx.de wrote:

> I will re-write the whole "directory" section on the wiki. For now I  
> will add the "group" reply.
>
> Thank you,
> Phil




From e.schmidbauer at gmail.com  Tue Dec 30 11:50:21 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 14:50:21 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
Message-ID: <2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>

i did as your said and got some errors in the console....
after i run pa call sip:886 at ww2.bwrl.org:5080 i get.....
2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
relaxxplayer->sip:886 at ww2.bwrl.org:5080 in context default
2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid
Profile
2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close
Channel N/A [CS_NEW]
2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate()
Cannot create outgoing channel of type [sofia] cause:
[DESTINATION_OUT_OF_ORDER]
2008-12-30 14:45:25 [INFO] mod_dptools.c:1891 audio_bridge_function()
Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER

any suggestions?


On Tue, Dec 30, 2008 at 2:06 PM, Brian West <brian at freeswitch.org> wrote:

> OK here try this..
> in portaudio.conf.xml
>
>     <param name="sample-rate" value="48000"/>
>
>
>     <param name="codec-ms" value="10"/>
>
>
> in dialplan/default.xml
>
>     <extension name="sip_uri">
>
>
>       <condition field="destination_number" expression="^sip:(.*)$">
>
>
>
>
>
>         <action application="bridge"
> data="{absolute_codec_string=CELT at 48000h@10i}sofia/${use_profile}/sip:$1"/>
>
>       </condition>
>
>
>     </extension>
>
> save that
> then
>
> pa call sip:886 at taz.bkw.org:5080
>
> /b
>
>
> On Dec 30, 2008, at 12:46 PM, e schmidbauer wrote:
>
> i have port audio setup but when i do a 'pa call <extension>' it enters the
> conference using the L16 codec. is there a way to use celt codec instead of
> the L16?
>
> On Tue, Dec 30, 2008 at 1:36 PM, Brian West <brian at freeswitch.org> wrote:
>
>> http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>
>> /b
>>
>> On Dec 30, 2008, at 12:33 PM, e schmidbauer wrote:
>>
>> > Could you explain in a more detail how you set that up?
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Dec 30 12:01:17 2008
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Dec 2008 12:01:17 -0800
Subject: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
	Available!
In-Reply-To: <B95277C3-C7E9-43A7-B4F4-6B55B7BD17CE@myrvold.org>
References: <20081230040049.GA2409@jdc.jasonjgw.net>
	<C57F2289.4B7B7%krice@suspicious.org>
	<87f2f3b90812292324r1c070458sb20c96db5df2be53@mail.gmail.com>
	<B95277C3-C7E9-43A7-B4F4-6B55B7BD17CE@myrvold.org>
Message-ID: <87f2f3b90812301201h2a0b06ebu3acaf72c15c0817d@mail.gmail.com>

Thanks for the heads up. I was able to confirm this behavior. I changed the
wiki to read "/noevents".
-MC

On Tue, Dec 30, 2008 at 3:03 AM, Ivan C Myrvold <ivan at myrvold.org> wrote:

> I found out that both "/event" and "/events" worked as commands, but only
> "/noevents" worked, not "/noevent", although the Wiki says "/noevent".
> Ivan
>
> Den 30. des.. 2008 kl. 08:24 skrev Michael Collins:
>
> Ken,
>
> Thanks for the clarification. I will make a note of this in the wiki. Also,
> can you hum a few bars and tell us what the "/filter" command does?
>
> -MC
>
> On Mon, Dec 29, 2008 at 10:57 PM, Ken Rice <krice at suspicious.org> wrote:
>
>> You can also use the ... Command to exit fs_cli and there are a few more
>> commands that are locally processed. (note: on the FS main console ...
>> Will
>> cause fs to shutdown. Fs_cli interprets this locally and it does not shut
>> down the main system. You stll need to do fsclt shutdown or something
>> similar)
>>
>> See the wiki for more information M Collins did a pretty good job
>> documenting it
>>
>> K
>>
>>
>> > From: Jason White <jason at jasonjgw.net>
>> > Reply-To: <freeswitch-users at lists.freeswitch.org>
>> > Date: Tue, 30 Dec 2008 15:00:49 +1100
>> > To: <freeswitch-users at lists.freeswitch.org>
>> > Subject: Re: [Freeswitch-users] FreeSWITCH News: New FreeSWITCH Client
>> > Available!
>> >
>> > By the way, the command to exit fs_cli is /exit (or /bye or /quit).
>> >
>> > Commands starting with / are handled internally by the process_command()
>> > function of the CLI, instead of being treated as FreeSWITCH API
>> commands.
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Dec 30 12:05:49 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 14:05:49 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
Message-ID: <50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>

Yes actually call sip:886 at taz.bkw.org:5080   I set it up specifically  
for you to test :P

/b

On Dec 30, 2008, at 1:50 PM, e schmidbauer wrote:

> i did as your said and got some errors in the console....
> after i run pa call sip:886 at ww2.bwrl.org:5080 i get.....
> 2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()  
> Processing relaxxplayer->sip:886 at ww2.bwrl.org:5080 in context default
> 2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel()  
> Invalid Profile
> 2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540  
> sofia_outgoing_channel() Close Channel N/A [CS_NEW]
> 2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116  
> switch_ivr_originate() Cannot create outgoing channel of type  
> [sofia] cause: [DESTINATION_OUT_OF_ORDER]
> 2008-12-30 14:45:25 [INFO] mod_dptools.c:1891  
> audio_bridge_function() Originate Failed.  Cause:  
> DESTINATION_OUT_OF_ORDER
>
> any suggestions?




From e.schmidbauer at gmail.com  Tue Dec 30 12:16:58 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 15:16:58 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
Message-ID: <2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>

tried  'pa call sip:886 at taz.bkw.org:5080'

2008-12-30 15:15:42 [NOTICE] switch_channel.c:565 switch_channel_set_name()
New Channel portaudio/sip:886 at taz.bkw.org:5080[a220653c-d6ae-11dd-af32-1ff260d7b236]
2008-12-30 15:15:42 [NOTICE] mod_portaudio.c:1586 place_call() Channel
[portaudio/sip:886 at taz.bkw.org:5080] has been answered
2008-12-30 15:15:42 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
FreeSwitch->sip:886 at taz.bkw.org:5080 in context default
2008-12-30 15:15:42 [ERR] mod_sofia.c:2423 sofia_outgoing_channel() Invalid
Profile
2008-12-30 15:15:42 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close
Channel N/A [CS_NEW]
2008-12-30 15:15:42 [ERR] switch_ivr_originate.c:1116 switch_ivr_originate()
Cannot create outgoing channel of type [sofia] cause:
[DESTINATION_OUT_OF_ORDER]
2008-12-30 15:15:42 [INFO] mod_dptools.c:1891 audio_bridge_function()
Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER
2008-12-30 15:15:42 [NOTICE] mod_dptools.c:1918 audio_bridge_function()
Hangup portaudio/sip:886 at taz.bkw.org:5080 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]

still got errors....


On Tue, Dec 30, 2008 at 3:05 PM, Brian West <brian at freeswitch.org> wrote:

> Yes actually call sip:886 at taz.bkw.org:5080   I set it up specifically
> for you to test :P
>
> /b
>
> On Dec 30, 2008, at 1:50 PM, e schmidbauer wrote:
>
> > i did as your said and got some errors in the console....
> > after i run pa call sip:886 at ww2.bwrl.org:5080 i get.....
> > 2008-12-30 14:45:25 [INFO] mod_dialplan_xml.c:233 dialplan_hunt()
> > Processing relaxxplayer->sip:886 at ww2.bwrl.org:5080 in context default
> > 2008-12-30 14:45:25 [ERR] mod_sofia.c:2423 sofia_outgoing_channel()
> > Invalid Profile
> > 2008-12-30 14:45:25 [NOTICE] mod_sofia.c:2540
> > sofia_outgoing_channel() Close Channel N/A [CS_NEW]
> > 2008-12-30 14:45:25 [ERR] switch_ivr_originate.c:1116
> > switch_ivr_originate() Cannot create outgoing channel of type
> > [sofia] cause: [DESTINATION_OUT_OF_ORDER]
> > 2008-12-30 14:45:25 [INFO] mod_dptools.c:1891
> > audio_bridge_function() Originate Failed.  Cause:
> > DESTINATION_OUT_OF_ORDER
> >
> > any suggestions?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Dec 30 12:32:11 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 14:32:11 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
Message-ID: <DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>

You don't have the default config ... did you modify the dialplan to  
have the propler sip_uri extension.. you only had to change one line.


On Dec 30, 2008, at 2:16 PM, e schmidbauer wrote:

> 2008-12-30 15:15:42 [ERR] mod_sofia.c:2423 sofia_outgoing_channel()  
> Invalid Profile
> 2008-12-30 15:15:42 [NOTICE] mod_sofia.c:2540  
> sofia_outgoing_channel() Close Channel N/A [CS_NEW]




From jaugenstine at gmail.com  Tue Dec 30 12:36:40 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Tue, 30 Dec 2008 12:36:40 -0800
Subject: [Freeswitch-users]  LUA execute response question
Message-ID: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>

I am developing a Freeswitch/LUA script.  From the script, I have a
requirement to retrieve information via HTTP from an app server.  I have
utilized the HTTP application from the Freeswitch CLI.  It works great.  My
question is how can I call the "http get http://www......." from within the
LUA script and retrieve the HTTP response?  Is this feasible?  Or is there a
way to make an HTTP request directly from LUA?

Jonathan
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From e.schmidbauer at gmail.com  Tue Dec 30 12:43:22 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 15:43:22 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
	<DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
Message-ID: <2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>

success! i was able to connect to your box using the CELT codec. thanks for
your help.
could you tell me how your end is configured so i can try this on my box?

On Tue, Dec 30, 2008 at 3:32 PM, Brian West <brian at freeswitch.org> wrote:

> You don't have the default config ... did you modify the dialplan to
> have the propler sip_uri extension.. you only had to change one line.
>
>
> On Dec 30, 2008, at 2:16 PM, e schmidbauer wrote:
>
> > 2008-12-30 15:15:42 [ERR] mod_sofia.c:2423 sofia_outgoing_channel()
> > Invalid Profile
> > 2008-12-30 15:15:42 [NOTICE] mod_sofia.c:2540
> > sofia_outgoing_channel() Close Channel N/A [CS_NEW]
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Dec 30 12:47:28 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 14:47:28 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<816E569A-7173-4564-B206-AD8F9A8371EC@freeswitch.org>
	<2cef777b0812301033r4a536073pbfebfadd469a7d0d@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
	<DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
	<2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>
Message-ID: <041099B7-5F37-4A65-9C4E-E20776094410@freeswitch.org>

Its setup to answer and playback... with CELT at 48000h allowed on the  
sofia profile.. playing a stream from my DirecTV (XM Radio 20on20 @ 48k)

btw how did it sound?

/b

On Dec 30, 2008, at 2:43 PM, e schmidbauer wrote:

> success! i was able to connect to your box using the CELT codec.  
> thanks for your help.
> could you tell me how your end is configured so i can try this on my  
> box?




From e.schmidbauer at gmail.com  Tue Dec 30 12:54:26 2008
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Tue, 30 Dec 2008 15:54:26 -0500
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <041099B7-5F37-4A65-9C4E-E20776094410@freeswitch.org>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
	<DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
	<2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>
	<041099B7-5F37-4A65-9C4E-E20776094410@freeswitch.org>
Message-ID: <2cef777b0812301254v77d3c2e5iaf9e8af400fdd148@mail.gmail.com>

sound quality is amazing like im listening to music on my own computer. can
you show me the dialplan config for the extension?

On Tue, Dec 30, 2008 at 3:47 PM, Brian West <brian at freeswitch.org> wrote:

> Its setup to answer and playback... with CELT at 48000h allowed on the
> sofia profile.. playing a stream from my DirecTV (XM Radio 20on20 @ 48k)
>
> btw how did it sound?
>
> /b
>
> On Dec 30, 2008, at 2:43 PM, e schmidbauer wrote:
>
> > success! i was able to connect to your box using the CELT codec.
> > thanks for your help.
> > could you tell me how your end is configured so i can try this on my
> > box?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Tue Dec 30 12:57:19 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 15:57:19 -0500
Subject: [Freeswitch-users] LUA execute response question
In-Reply-To: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
References: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
Message-ID: <EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>

you can execute the freeswitch api command like any other api command  
or use any loadable lua module available to do this.


Mike

On Dec 30, 2008, at 3:36 PM, jonathan augenstine wrote:

> I am developing a Freeswitch/LUA script.  From the script, I have a  
> requirement to retrieve information via HTTP from an app server.  I  
> have utilized the HTTP application from the Freeswitch CLI.  It  
> works great.  My question is how can I call the "http get http:// 
> www......." from within the LUA script and retrieve the HTTP  
> response?  Is this feasible?  Or is there a way to make an HTTP  
> request directly from LUA?

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From brian at freeswitch.org  Tue Dec 30 12:59:43 2008
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Dec 2008 14:59:43 -0600
Subject: [Freeswitch-users] how to use celt codec
In-Reply-To: <2cef777b0812301254v77d3c2e5iaf9e8af400fdd148@mail.gmail.com>
References: <2cef777b0812300956t4d79b300sa224fc9526f58587@mail.gmail.com>
	<B6530DFE-AE18-43D2-9201-C3445550CDD5@freeswitch.org>
	<2cef777b0812301046v2a30a770v2dc8f83313fefd22@mail.gmail.com>
	<A9194FEE-9AC5-4C42-A382-74A87B9605C2@freeswitch.org>
	<2cef777b0812301150w4f51f230j93a1962744ab2b73@mail.gmail.com>
	<50208100-7188-45B1-94E2-C264235F2179@freeswitch.org>
	<2cef777b0812301216g48341ceah7aef9f9be1c11a2f@mail.gmail.com>
	<DD95F1C7-8AD1-4C4D-8F38-8DFC7DFA703B@freeswitch.org>
	<2cef777b0812301243j394e5e76yd6db7f8421c7a940@mail.gmail.com>
	<041099B7-5F37-4A65-9C4E-E20776094410@freeswitch.org>
	<2cef777b0812301254v77d3c2e5iaf9e8af400fdd148@mail.gmail.com>
Message-ID: <E391462D-CA01-488C-A923-15B02674F86A@freeswitch.org>

     <extension name="xmradio">
       <condition field="destination_number" expression="^886$">
         <action application="answer"/>
         <action application="playback" data="local_stream://xmradio"/>
       </condition>
     </extension>


and local stream is just pointed at a shout cast server on my Mac that  
is plugged into my DirecTV receiver

/b


On Dec 30, 2008, at 2:54 PM, e schmidbauer wrote:

> sound quality is amazing like im listening to music on my own  
> computer. can you show me the dialplan config for the extension?




From jaugenstine at gmail.com  Tue Dec 30 13:16:36 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Tue, 30 Dec 2008 13:16:36 -0800
Subject: [Freeswitch-users] LUA execute response question
In-Reply-To: <EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>
References: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
	<EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>
Message-ID: <207e7a5e0812301316k2c7ba5dft5ba7e1c51795301a@mail.gmail.com>

Mike,

I executed the following:

>From CLI:
http get http://www.google.com {}

This comes back with the HTTP response and prints it to the console.  I
tried to execute the following:

response = session:execute("http", "get http://www.google.com {}");

What I am unable to figure out how to do is retrieve HTTP response.  That is
my dilemma.

Jonathan

On Tue, Dec 30, 2008 at 12:57 PM, Michael Jerris <mike at jerris.com> wrote:

> you can execute the freeswitch api command like any other api command or
> use any loadable lua module available to do this.
>
> Mike
>
> On Dec 30, 2008, at 3:36 PM, jonathan augenstine wrote:
>
> I am developing a Freeswitch/LUA script.  From the script, I have a
> requirement to retrieve information via HTTP from an app server.  I have
> utilized the HTTP application from the Freeswitch CLI.  It works great.  My
> question is how can I call the "http get http://www......." from within
> the LUA script and retrieve the HTTP response?  Is this feasible?  Or is
> there a way to make an HTTP request directly from LUA?
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Dec 30 13:40:45 2008
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Dec 2008 16:40:45 -0500
Subject: [Freeswitch-users] LUA execute response question
In-Reply-To: <207e7a5e0812301316k2c7ba5dft5ba7e1c51795301a@mail.gmail.com>
References: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
	<EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>
	<207e7a5e0812301316k2c7ba5dft5ba7e1c51795301a@mail.gmail.com>
Message-ID: <413D08F3-9803-450A-B9B8-EE8A7F965CBF@jerris.com>

session:execute runs a freeswitch app, you want to be executing the  
fsapi command using API:execute
Mike


On Dec 30, 2008, at 4:16 PM, jonathan augenstine wrote:

> Mike,
>
> I executed the following:
>
> From CLI:
> http get http://www.google.com {}
>
> This comes back with the HTTP response and prints it to the  
> console.  I tried to execute the following:
>
> response = session:execute("http", "get http://www.google.com {}");
>
> What I am unable to figure out how to do is retrieve HTTP response.   
> That is my dilemma.
>
> Jonathan
>
> On Tue, Dec 30, 2008 at 12:57 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> you can execute the freeswitch api command like any other api  
> command or use any loadable lua module available to do this.
>
>
> Mike
>
> On Dec 30, 2008, at 3:36 PM, jonathan augenstine wrote:
>
>> I am developing a Freeswitch/LUA script.  From the script, I have a  
>> requirement to retrieve information via HTTP from an app server.  I  
>> have utilized the HTTP application from the Freeswitch CLI.  It  
>> works great.  My question is how can I call the "http get http:// 
>> www......." from within the LUA script and retrieve the HTTP  
>> response?  Is this feasible?  Or is there a way to make an HTTP  
>> request directly from LUA?

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From dyfet at gnutelephony.org  Tue Dec 30 13:48:05 2008
From: dyfet at gnutelephony.org (David Sugar)
Date: Tue, 30 Dec 2008 16:48:05 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar - a
 cute hack
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670233BC664C@mse17be1.mse17.exchange.ms>
References: <495A124C.3040006@laposte.net>
	<495A1D2D.3070507@gnutelephony.org>	<495A22E0.3040904@networkoblivion.com>
	<6E8D2069C08AA84A83D336E996AE4C670233BC664C@mse17be1.mse17.exchange.ms>
Message-ID: <495A9715.4020201@gnutelephony.org>

I actually have found an alternate approach that we optionally use in
sipwitch.  Basically, sipwitch can be set to recognize a "trusted"
subnet, and automatically accepts a refresh from any actively registered
ua on the trusted subnet(s) without requesting an authentication
challenge, so long as the ua refreshes from the same sip port and ip
address it originally registered and authenticated from.  It will also
do the same for invites and other otherwise "authentication challenge"
sip requests that can originate from ua's on the trusted subnet(s).

Using this option of course kills any ability to proxy register multiple
ua's through another sip server, although this can be solved by
recognizing certain id's as explicitly not trustable.  However, for most
common configurations and use cases, it works very well and does
effectively halve sip network traffic :).

Michael Giagnocavo wrote:
>>>  This is
>>> because the ua sends it's registration refresh unauthenticated.  The
>>> registrar will then push back an authentication challenge request so the
>>> ua can prove its identity, at which point the ua then repeats the same
>>> transaction, but with authentication credentials attached.
>> Why does it do that?  Every time I do a debug, I see the first request
>> denied as unauthorized and then it always comes right back and gets
> 
> Welcome to HTTP Digest authentication. The request has to get challenged to get a new nonce from the server (so as to mitigate replay attacks).
> 
> You could TLS and auth off of the client cert, except few devices support that, and you'd have the "overhead" of TCP (which is like bad or something).
> 
> -Michael
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dyfet at gnutelephony.org  Tue Dec 30 13:50:30 2008
From: dyfet at gnutelephony.org (David Sugar)
Date: Tue, 30 Dec 2008 16:50:30 -0500
Subject: [Freeswitch-users] Freeswitch optimization as a registrar
In-Reply-To: <7b197bef0812300527x5073b212j38b1a60f475440f6@mail.gmail.com>
References: <495A124C.3040006@laposte.net> <495A1D2D.3070507@gnutelephony.org>
	<7b197bef0812300527x5073b212j38b1a60f475440f6@mail.gmail.com>
Message-ID: <495A97A6.1060504@gnutelephony.org>

Well, there are worse virus's one could be infected with, I suppose ;).

Actually recently I had been surviving focusing on secure VoIP and
wireless...

Giovanni Maruzzelli wrote:
> Hi David,
> 
> very happy to read you on the FS list!
> 
> We met in 2001 at OSCon San Diego, where you "infected" me with the
> telephony virus :-).
> 
> You did great work with the Bayonne project, really breaking new ground.
> 
> Thank you,
> 
> happy hacking,
> 
> happy new year!!!!
> 
> 
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> =========================================
> Company : Celliax
> Website: www.celliax.org
> Address : via Pierlombardo 9, 20135 Milano
> Country/Territory : Italy
> Business Email: gmaruzz at celliax dot org
> Cell : 39-347-2665618
> Fax : 39-02-87390039
> 
> 
> 
> 
> On Tue, Dec 30, 2008 at 2:07 PM, David Sugar <dyfet at gnutelephony.org> wrote:
>> You actually have potentially ~1320 effective "SIP transactions" per
>> second to support 40000 registered ua's with a 60s refresh.  This is
>> because the ua sends it's registration refresh unauthenticated.  The
>> registrar will then push back an authentication challenge request so the
>> ua can prove its identity, at which point the ua then repeats the same
>> transaction, but with authentication credentials attached.
>>
>> rod wrote:
>>> Hi all,
>>>
>>> I know that freeswitch has not been designed as a pure sip
>>> proxy/registrar, but I'm wondering how many subscribers could be handled
>>> by FS.
>>>
>>> I setup the following test environment:
>>>     - Kamailio 1.4.2 as the registrar
>>>     - all invite requests are flowing through FS, even for a call
>>> between 2 registered subscribers. Many reasons for this: the calls CDR
>>> are centralized in the same format, I can easily add a billing ID to a
>>> call, proceed to recording, set the caller as anonymous if requested...
>>>     - FS is used also as a SBC
>>>
>>> There is still a lot of work to do, mainly on the call forwarding
>>> feature and this is why I'm wondering (simply out of curiosity) what
>>> could have been achieved using only FS (easier to setup when only one
>>> equipment is involved :) ).
>>>
>>> I'd like to register 40 000 subscribers (if each user registers every
>>> 60s, you have approx 670 registration per second, this setup is working
>>> on Kamailio).
>>>
>>> I did the following to increase FS performance regarding registration:
>>>     - put the directory containing users in a RAMDISK
>>>     - put the db directory in a RAMDISK
>>>
>>> with this I was able to reach 190 registration per second (50 without
>>> the ramdisk) but for one SIP account, not too useful :p (for your
>>> information I see a huge improvement when switching from 1.0.1 phoenix:
>>> 150cps to FS svn 105xx: 190)
>>> When trying with 25000 SIP accounts, I got no more than 30cps.
>>>
>>> Then I tried to use the odbc mysql for registration, using this I was
>>> able to achieve 50cps. The mysql DB is not in a RAMDISK. For all these
>>> tests, the presence support has been disabled.
>>>
>>> As the IO performance seems to be a bottleneck, I'd like to know if
>>> there is a way to store the registration in memory only without database
>>> persistency.
>>>
>>> This thread is there only to share tips, not to complain about FS poor
>>> performance as a SIP registrar when compared to Kamailio. If I compare
>>> FS to a commercial SBC I'm using in production, I have to say that FS is
>>> really a great piece of software (lacks only statistics module, snmp,
>>> and heartbeat redundancy for failover).
>>>
>>> regards,
>>> rod
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jaugenstine at gmail.com  Tue Dec 30 15:31:07 2008
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Tue, 30 Dec 2008 15:31:07 -0800
Subject: [Freeswitch-users] LUA execute response question
In-Reply-To: <413D08F3-9803-450A-B9B8-EE8A7F965CBF@jerris.com>
References: <207e7a5e0812301236v1ef45264g69419b0755716e0a@mail.gmail.com>
	<EF49F5F5-332F-4EB3-AA71-08DD7DA72021@jerris.com>
	<207e7a5e0812301316k2c7ba5dft5ba7e1c51795301a@mail.gmail.com>
	<413D08F3-9803-450A-B9B8-EE8A7F965CBF@jerris.com>
Message-ID: <207e7a5e0812301531x7cdff10bmf59c0c2e79381814@mail.gmail.com>

Mike,

Thank you.  I was confused and it is working now.

Jonathan


On Tue, Dec 30, 2008 at 1:40 PM, Michael Jerris <mike at jerris.com> wrote:

> session:execute runs a freeswitch app, you want to be executing the fsapi
> command using API:executeMike
>
>
> On Dec 30, 2008, at 4:16 PM, jonathan augenstine wrote:
>
> Mike,
>
> I executed the following:
>
> From CLI:
> http get http://www.google.com {}
>
> This comes back with the HTTP response and prints it to the console.  I
> tried to execute the following:
>
> response = session:execute("http", "get http://www.google.com {}");
>
> What I am unable to figure out how to do is retrieve HTTP response.  That
> is my dilemma.
>
> Jonathan
>
> On Tue, Dec 30, 2008 at 12:57 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> you can execute the freeswitch api command like any other api command or
>> use any loadable lua module available to do this.
>>
>> Mike
>>
>> On Dec 30, 2008, at 3:36 PM, jonathan augenstine wrote:
>>
>> I am developing a Freeswitch/LUA script.  From the script, I have a
>> requirement to retrieve information via HTTP from an app server.  I have
>> utilized the HTTP application from the Freeswitch CLI.  It works great.  My
>> question is how can I call the "http get http://www......." from within
>> the LUA script and retrieve the HTTP response?  Is this feasible?  Or is
>> there a way to make an HTTP request directly from LUA?
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fvillarroel at yahoo.com  Tue Dec 30 18:10:59 2008
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Tue, 30 Dec 2008 18:10:59 -0800 (PST)
Subject: [Freeswitch-users] New
Message-ID: <196421.77116.qm@web34307.mail.mud.yahoo.com>

Dear All,

I am new on this list, i am chilean.

I come from Asterisk and my company is dedicated to wholesale, i am provider for chile mobile.

We will need implement FreeSwictch before Asterisk for Retail and Wholesale

How i will can starting, please recomended any how to or pdf for beginning.

Is possible administrator FS from any GUI, but no from Centos Image FS,I like Debian. 

Other questions, FreeSwitch is more complex that Asterisk for begining?

Thanks everyone and excuse me English.

Fernando



      



From jason at jasonjgw.net  Tue Dec 30 18:43:48 2008
From: jason at jasonjgw.net (Jason White)
Date: Wed, 31 Dec 2008 13:43:48 +1100
Subject: [Freeswitch-users] New
In-Reply-To: <196421.77116.qm@web34307.mail.mud.yahoo.com>
References: <196421.77116.qm@web34307.mail.mud.yahoo.com>
Message-ID: <20081231024348.GA10230@jdc.jasonjgw.net>

FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:
 
> How i will can starting, please recomended any how to or pdf for beginning.

http://wiki.freeswitch.org/
> 
> 
> Other questions, FreeSwitch is more complex that Asterisk for begining?

No, just different from asterisk.




From intralanman at freeswitch.org  Wed Dec 31 06:19:52 2008
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 31 Dec 2008 14:19:52 +0000
Subject: [Freeswitch-users] New
In-Reply-To: <196421.77116.qm@web34307.mail.mud.yahoo.com>
References: <196421.77116.qm@web34307.mail.mud.yahoo.com>
Message-ID: <495B7F88.3010301@freeswitch.org>

FERNANDO VILLARROEL wrote:
> Dear All,
>
> I am new on this list, i am chilean.
>
> I come from Asterisk and my company is dedicated to wholesale, i am provider for chile mobile.
>
> We will need implement FreeSwictch before Asterisk for Retail and Wholesale
>
> How i will can starting, please recomended any how to or pdf for beginning.
>   
A couple of good starting points are:
http://wiki.freeswitch.org/wiki/Installation_Guide
http://wiki.freeswitch.org/wiki/Getting_Started_Guide

Since those are written in only english at the present time, you might 
want to join the irc channel, it's #freeswitch on irc.freenode.net ... 
there are lots of languages spoken in the channel, so someone could 
probably help you more than those guides.
> Is possible administrator FS from any GUI, but no from Centos Image FS,I like Debian. 
>   
Right now there are no GUIs completed, but several being worked on.
> Other questions, FreeSwitch is more complex that Asterisk for begining?
>   
Depending on your background, you might find it harder, or you might 
find it easier... FreeSWITCH configs are XML based where Asterisk 
configs are INI based. That's only the start of the differences.. I'd 
say give them both a try and see which shoe fits best :-)
> Thanks everyone and excuse me English.
>   
Your english isn't that bad, thanks for posting.

-Ray



From Prometheus001 at gmx.net  Wed Dec 31 06:28:32 2008
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 31 Dec 2008 15:28:32 +0100
Subject: [Freeswitch-users] uuid_playback
Message-ID: <495B8190.2080207@gmx.net>

As I see on the Wiki page uuid_playback seems to be implemented, however
it doesn't work on the console or via event_socket.
Also in the code I could not find it (svn 10438).

So for now I use uuid_brodcast to play announcements to one or both parties.

Question: What is the status of uuid_playback?

Best regards
Peter




From gmaruzz at celliax.org  Wed Dec 31 06:58:08 2008
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 31 Dec 2008 15:58:08 +0100
Subject: [Freeswitch-users] mod_skypiax inching forward
Message-ID: <7b197bef0812310658u2cea5f0p7e81ea099af9de83@mail.gmail.com>

Hi FreeSWITCHers!

mod_skypiax, the Skype compatible endpoint, is slowly inching toward release :-)

When the demo is online (will go on and off for development), you can
test it (so helping finding bugs) by calling with Skype the Skype
Names:

skypiax20, skypiax19, skypiax18, ...., skypiax1

Happy New Year !!!

Sincerely,

Giovanni Maruzzelli
=========================================
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039



From javieraristizabal at gmail.com  Wed Dec 31 08:57:33 2008
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Wed, 31 Dec 2008 11:57:33 -0500
Subject: [Freeswitch-users] New
In-Reply-To: <196421.77116.qm@web34307.mail.mud.yahoo.com>
References: <196421.77116.qm@web34307.mail.mud.yahoo.com>
Message-ID: <b6f11f300812310857w38c01d9fy5345ea32d647ac23@mail.gmail.com>

Hola Fernando, llevo algun tiempo trabajando con Freeswitch, si algo te
puedo ayudar me lo puedes comentar. A traves de este medio o a traves del
IRC mi nick es "javar".
Felices Fiestas.


Javier.
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From kristian.kielhofner at gmail.com  Wed Dec 31 10:44:56 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 13:44:56 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was Sonus
	and RFC 2833 related
Message-ID: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>

Hey everyone,

  A few days ago I was reading about a bug with Sonus gear and
FreeSwitch.  I remember it because Tony implemented a workaround the
buggy Sonus gear based on the SDP, all while adding a snarky (yet
hilarious) log message.

  You guessed it, I'm having 2833 timestamp problems with Sonus...

  I can't find this issue for the life of me in Jira.  Does anyone
know where it is?  It was fixed in rev 10744 with the commit message
"sonus, sonus, sonus sonus is a four letter word".

Thanks!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From brian at freeswitch.org  Wed Dec 31 11:23:23 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 31 Dec 2008 13:23:23 -0600
Subject: [Freeswitch-users] uuid_playback
In-Reply-To: <495B8190.2080207@gmx.net>
References: <495B8190.2080207@gmx.net>
Message-ID: <F952536A-A68B-4F3C-A670-0F005201FAAC@freeswitch.org>

Wiki link please.

/b

On Dec 31, 2008, at 8:28 AM, Peter P GMX wrote:

> As I see on the Wiki page uuid_playback seems to be implemented,  
> however
> it doesn't work on the console or via event_socket.
> Also in the code I could not find it (svn 10438).
>
> So for now I use uuid_brodcast to play announcements to one or both  
> parties.
>
> Question: What is the status of uuid_playback?
>
> Best regards
> Peter




From brian at freeswitch.org  Wed Dec 31 11:24:46 2008
From: brian at freeswitch.org (Brian West)
Date: Wed, 31 Dec 2008 13:24:46 -0600
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
Message-ID: <52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>

I would recommend getting the latest SVN since we had to break out the  
cisco and sonus rtp issues... Try this http://wiki.freeswitch.org/wiki/RTP_Issues

/b

On Dec 31, 2008, at 12:44 PM, Kristian Kielhofner wrote:

> Hey everyone,
>
>  A few days ago I was reading about a bug with Sonus gear and
> FreeSwitch.  I remember it because Tony implemented a workaround the
> buggy Sonus gear based on the SDP, all while adding a snarky (yet
> hilarious) log message.
>
>  You guessed it, I'm having 2833 timestamp problems with Sonus...
>
>  I can't find this issue for the life of me in Jira.  Does anyone
> know where it is?  It was fixed in rev 10744 with the commit message
> "sonus, sonus, sonus sonus is a four letter word".
>
> Thanks!




From kristian.kielhofner at gmail.com  Wed Dec 31 12:14:04 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 15:14:04 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
Message-ID: <2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>

On Wed, Dec 31, 2008 at 2:24 PM, Brian West <brian at freeswitch.org> wrote:
> I would recommend getting the latest SVN since we had to break out the
> cisco and sonus rtp issues... Try this http://wiki.freeswitch.org/wiki/RTP_Issues
>
> /b
>

Brian,

  Thanks, I have and I was well aware of these (I've been lurking).

  However, my issue is with another platform and Sonus.  I want to
demonstrate that this is a *known* issue with Sonus gear.  Plus, I
found it:

http://jira.freeswitch.org/browse/FSCORE-251

Thanks again!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From mike at jerris.com  Wed Dec 31 12:33:37 2008
From: mike at jerris.com (Michael Jerris)
Date: Wed, 31 Dec 2008 15:33:37 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
Message-ID: <43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>

If your looking for specifics of where they are being stupid and/or  
violating rfc's pop on by and let us know, we can probably detail some  
stuff that is not in the bugs as well.

Mike

On Dec 31, 2008, at 3:14 PM, Kristian Kielhofner wrote:

> On Wed, Dec 31, 2008 at 2:24 PM, Brian West <brian at freeswitch.org>  
> wrote:
>> I would recommend getting the latest SVN since we had to break out  
>> the
>> cisco and sonus rtp issues... Try this http://wiki.freeswitch.org/wiki/RTP_Issues
>>
>> /b
>>
>
> Brian,
>
>  Thanks, I have and I was well aware of these (I've been lurking).
>
>  However, my issue is with another platform and Sonus.  I want to
> demonstrate that this is a *known* issue with Sonus gear.  Plus, I
> found it:
>
> http://jira.freeswitch.org/browse/FSCORE-251
>
> Thanks again!
>
> -- 
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From kristian.kielhofner at gmail.com  Wed Dec 31 13:01:10 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 16:01:10 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
	<43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>
Message-ID: <2d9149cd0812311301h2b632e45g5534aba7011c809f@mail.gmail.com>

On 12/31/08, Michael Jerris <mike at jerris.com> wrote:
> If your looking for specifics of where they are being stupid and/or
>  violating rfc's pop on by and let us know, we can probably detail some
>  stuff that is not in the bugs as well.
>
>  Mike

Mike,

  Thanks but I (think) I basically figured it out...

  I am using G.729 and RFC 2833 DTMF.

  If you send a 2833 event and voice packet with the same timestamp
Sonus will drop the event (maybe both) EVEN IF they have different
sequence numbers.  Nevermind that this behavior (same timestamps) is
quite possible and even desired in some cases and violates both RFC
1889/3550 and RFC 2833/4733.

  I didn't dig into the code but it looks like the Freeswitch
workaround just tweaks the timestamps to get around this if a Sonus
originator is detected (by parsing the SDP).  Nice.

  Did I get it?

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From anthony.minessale at gmail.com  Wed Dec 31 13:02:05 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Dec 2008 15:02:05 -0600
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
Message-ID: <191c3a030812311302w2545d099la837885e37ee412d@mail.gmail.com>

In case you want to know the nitty gritty.

excerpt from switch_types.h

RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833 = (1 << 1)
    /*
      Sonus wrongly expects that, when sending a multi-packet 2833 DTMF
event, The sender
      should increment the RTP timestamp in each packet when, in reality,
the sender should
      send the same exact timestamp and increment the duration field in the
2833 payload.
      This allows a reconstruction of the duration if any of the packets are
lost.

      final_duration - initial_timestamp = total_samples

      However, if the duration value exceeds the space allocated (16 bits),
The sender should increment
      the timestamp one unit and reset the duration to 0.

      Always sending a duration of 0 with a new timestamp should be
tolerated but is rarely intentional
      and is mistakenly done by many devices.
      The issue is that the Sonus expects everyone to do it this way instead
of tolerating either way.
      Sonus will actually ignore every packet with the same timestamp before
concluding if it's DTMF.

      This flag will cause each packet to have a new timestamp.
    */



On Wed, Dec 31, 2008 at 2:14 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On Wed, Dec 31, 2008 at 2:24 PM, Brian West <brian at freeswitch.org> wrote:
> > I would recommend getting the latest SVN since we had to break out the
> > cisco and sonus rtp issues... Try this
> http://wiki.freeswitch.org/wiki/RTP_Issues
> >
> > /b
> >
>
> Brian,
>
>  Thanks, I have and I was well aware of these (I've been lurking).
>
>  However, my issue is with another platform and Sonus.  I want to
> demonstrate that this is a *known* issue with Sonus gear.  Plus, I
> found it:
>
> http://jira.freeswitch.org/browse/FSCORE-251
>
> Thanks again!
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristian.kielhofner at gmail.com  Wed Dec 31 13:06:49 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 16:06:49 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <191c3a030812311302w2545d099la837885e37ee412d@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
	<191c3a030812311302w2545d099la837885e37ee412d@mail.gmail.com>
Message-ID: <2d9149cd0812311306i13777b93y503852bd2386865d@mail.gmail.com>

On 12/31/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> In case you want to know the nitty gritty.
>
> excerpt from switch_types.h
>
> RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833 = (1 << 1)
>     /*
>       Sonus wrongly expects that, when sending a multi-packet 2833 DTMF
> event, The sender
>        should increment the RTP timestamp in each packet when, in reality,
> the sender should
>       send the same exact timestamp and increment the duration field in the
> 2833 payload.
>       This allows a reconstruction of the duration if any of the packets are
> lost.
>
>       final_duration - initial_timestamp = total_samples
>
>       However, if the duration value exceeds the space allocated (16 bits),
> The sender should increment
>       the timestamp one unit and reset the duration to 0.
>
>       Always sending a duration of 0 with a new timestamp should be
> tolerated but is rarely intentional
>       and is mistakenly done by many devices.
>       The issue is that the Sonus expects everyone to do it this way instead
> of tolerating either way.
>        Sonus will actually ignore every packet with the same timestamp
> before concluding if it's DTMF.
>
>       This flag will cause each packet to have a new timestamp.
>     */
>

Thanks Anthony!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From anthony.minessale at gmail.com  Wed Dec 31 13:15:28 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Dec 2008 15:15:28 -0600
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <2d9149cd0812311301h2b632e45g5534aba7011c809f@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
	<43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>
	<2d9149cd0812311301h2b632e45g5534aba7011c809f@mail.gmail.com>
Message-ID: <191c3a030812311315v4fa295d3rda7b1db1ce68b85d@mail.gmail.com>

in case you want to know the other big annoying one:

If you suddenly change timestamp base mid call, sonus will lose 2 sec of
audio trying to adjust.

Say you have an ivr that asks you to dial an ext then places the call.
While the ivr is the far end and interacting with the sonus there is a
series of timestamps generated by FS.
Then when the call is placed we start passing through the timestamps from
the new farther far end so the jitter can
be preserved.  (if we rewrite the timestamps to our original ones, any
jitter would be locked in place).

With sonus, you pretty much have to set FS to rewrite the timestamps or live
with the audio drop =(




On Wed, Dec 31, 2008 at 3:01 PM, Kristian Kielhofner <
kristian.kielhofner at gmail.com> wrote:

> On 12/31/08, Michael Jerris <mike at jerris.com> wrote:
> > If your looking for specifics of where they are being stupid and/or
> >  violating rfc's pop on by and let us know, we can probably detail some
> >  stuff that is not in the bugs as well.
> >
> >  Mike
>
> Mike,
>
>  Thanks but I (think) I basically figured it out...
>
>  I am using G.729 and RFC 2833 DTMF.
>
>  If you send a 2833 event and voice packet with the same timestamp
> Sonus will drop the event (maybe both) EVEN IF they have different
> sequence numbers.  Nevermind that this behavior (same timestamps) is
> quite possible and even desired in some cases and violates both RFC
> 1889/3550 and RFC 2833/4733.
>
>  I didn't dig into the code but it looks like the Freeswitch
> workaround just tweaks the timestamps to get around this if a Sonus
> originator is detected (by parsing the SDP).  Nice.
>
>  Did I get it?
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From kristian.kielhofner at gmail.com  Wed Dec 31 13:35:48 2008
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 31 Dec 2008 16:35:48 -0500
Subject: [Freeswitch-users] Looking for a specific bug in jira - was
	Sonus and RFC 2833 related
In-Reply-To: <191c3a030812311315v4fa295d3rda7b1db1ce68b85d@mail.gmail.com>
References: <2d9149cd0812311044u39f54127x8ec882fb1f15c27a@mail.gmail.com>
	<52342ABB-A32C-41D6-AAB6-38D4088491A5@freeswitch.org>
	<2d9149cd0812311214u76a98932l6908c072308abdc6@mail.gmail.com>
	<43970EFF-82F4-4114-8827-CB5CD6890F9C@jerris.com>
	<2d9149cd0812311301h2b632e45g5534aba7011c809f@mail.gmail.com>
	<191c3a030812311315v4fa295d3rda7b1db1ce68b85d@mail.gmail.com>
Message-ID: <2d9149cd0812311335k2ad2595ew2840f6076d37cb3c@mail.gmail.com>

On 12/31/08, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> in case you want to know the other big annoying one:
>
> If you suddenly change timestamp base mid call, sonus will lose 2 sec of
> audio trying to adjust.
>
> Say you have an ivr that asks you to dial an ext then places the call.
>  While the ivr is the far end and interacting with the sonus there is a
> series of timestamps generated by FS.
> Then when the call is placed we start passing through the timestamps from
> the new farther far end so the jitter can
>  be preserved.  (if we rewrite the timestamps to our original ones, any
> jitter would be locked in place).
>
> With sonus, you pretty much have to set FS to rewrite the timestamps or live
> with the audio drop =(
>

Anthony,

  Thanks for the pointer.  It's nice to also know that I won't be able
to implement *proper* jitter buffering in my network as long as Sonus
is involved somewhere in the call path.

  Sheesh.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com



From anthony.minessale at gmail.com  Wed Dec 31 14:47:11 2008
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Dec 2008 16:47:11 -0600
Subject: [Freeswitch-users] uuid_playback
In-Reply-To: <495B8190.2080207@gmx.net>
References: <495B8190.2080207@gmx.net>
Message-ID: <191c3a030812311447t59f1e093n143b50c7fa77ca3f@mail.gmail.com>

there is no such thing as uuid_playback
broadcast is the correct and only way besides maybe uuid_displace



On Wed, Dec 31, 2008 at 8:28 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> As I see on the Wiki page uuid_playback seems to be implemented, however
> it doesn't work on the console or via event_socket.
> Also in the code I could not find it (svn 10438).
>
> So for now I use uuid_brodcast to play announcements to one or both
> parties.
>
> Question: What is the status of uuid_playback?
>
> Best regards
> Peter
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From fvillarroel at yahoo.com  Wed Dec 31 16:37:55 2008
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Wed, 31 Dec 2008 16:37:55 -0800 (PST)
Subject: [Freeswitch-users] New
In-Reply-To: <b6f11f300812310857w38c01d9fy5345ea32d647ac23@mail.gmail.com>
Message-ID: <628881.28122.qm@web34304.mail.mud.yahoo.com>

Hello

Thanks all for your comments and specially to Javier and Raymond.

Ya me tendras haciendote consultas Javier.

Regards

Fernando


--- On Wed, 12/31/08, Javier Aristiz?bal <javieraristizabal at gmail.com> wrote:

> From: Javier Aristiz?bal <javieraristizabal at gmail.com>
> Subject: Re: [Freeswitch-users] New
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, December 31, 2008, 12:57 PM
> Hola Fernando, llevo algun tiempo trabajando con Freeswitch,
> si algo te
> puedo ayudar me lo puedes comentar. A traves de este medio
> o a traves del
> IRC mi nick es "javar".
> Felices Fiestas.
> 
> 
> Javier.
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


      



From jason at jasonjgw.net  Wed Dec 31 23:20:45 2008
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Jan 2009 18:20:45 +1100
Subject: [Freeswitch-users] fs_cli help command
Message-ID: <20090101072045.GA12582@jdc.jasonjgw.net>

I just noticed, as confirmed by reading the code, that now when the user types
"help" at the fs_cli prompt, the fs_cli help text is printed; but if what one
really wants is to execute the API help command, there doesn't seem to be any
way to do it.

process_command() gets the help command first, and there's no way to have it
passed to FreeSWITCH as an API command.

Here are a few options for solving this (I'm sure there are others):

1. Make the API command processing into a separate function, and have the
"help" command call it to execute "api help" after printing its own text. I
suppose one could also use a goto for this, but that could be accused of
inelegance.

2. Rename the fs_cli help command to "/help" - probably confusing to new
users!

3. Add a /help command that runs "api help".

I'm leaning toward option 3, but opinions may differ.