From sshen at oa.com.au  Sun May  3 21:00:56 2009
From: sshen at oa.com.au (Scott Shen)
Date: Mon, 4 May 2009 14:00:56 +1000
Subject: [Freeswitch-dev] originate creates a channel that can not be killed
Message-ID: <4462DF002246D44CB823FA4F985539941426D33C27@oa-exchange1.oa.com.au>

I am using FS r13096, found following problem


 1.  originate {originate_timeout=6,originate_retries=3,originate_retry_sleep_ms=60000}user/1002 &bridge(user/1007)
 2.  when 1002 side ringing, quickly pickup and hangup
 3.  show channels gives something like
31b614d0-999d-406e-bdba-b18ead0d26d8,outbound,2009-05-04 13:42:56,1241408576,sofia/internal/sip:1002 at 192.168.10.100:5060;line=xStcbdok,CS_EXECUTE,FreeSWITCH,0000000000,,sip:1002 at 192.168.10.100:5060;line=xStcbdok,bridge,user/1007,,default,PCMA,8000,PCMA,8000

 1.  when use uuid_kill of above uuid, FS reports "No Such Channel"
 2.  then at 60000ms interval, FS will give messages like 'Originate attempt x/3 in 60000ms'
 3.  when x reaches 1, the channel will be cleaned.

On step 3 and 4, if there's "No Such Channel", isn't that the channel should not be displayed?

On step 5, although the "originate" is attempted ( which shouldn't, because the call is already hangup on step 2 ), but neither caller nor the callee is ringed.

Also, is there a way to cancel "originate" ? eg send a cancel after invite.

Regards
Scott
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From anthony.minessale at gmail.com  Mon May  4 06:11:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 May 2009 08:11:37 -0500
Subject: [Freeswitch-dev] originate creates a channel that can not be
	killed
In-Reply-To: <4462DF002246D44CB823FA4F985539941426D33C27@oa-exchange1.oa.com.au>
References: <4462DF002246D44CB823FA4F985539941426D33C27@oa-exchange1.oa.com.au>
Message-ID: <191c3a030905040611r68fc3973ufb77a8cf83d5740f@mail.gmail.com>

Please do not report bugs on the mailing list.  It is impossible to manage
them from here.

The proper place to submit bug reports is

http://jira.freeswitch.org


On Sun, May 3, 2009 at 11:00 PM, Scott Shen <sshen at oa.com.au> wrote:

>  I am using FS r13096, found following problem
>
>
>
>    1. originate
>    {originate_timeout=6,originate_retries=3,originate_retry_sleep_ms=60000}user/1002
>    &bridge(user/1007)
>    2. when 1002 side ringing, quickly pickup and hangup
>    3. show channels gives something like
>
> 31b614d0-999d-406e-bdba-b18ead0d26d8,outbound,2009-05-04
> 13:42:56,1241408576,sofia/internal/sip:1002 at 192.168.10.100:5060
> ;line=xStcbdok,CS_EXECUTE,FreeSWITCH,0000000000,,sip:1002 at 192.168.10.100:5060
> ;line=xStcbdok,bridge,user/1007,,default,PCMA,8000,PCMA,8000
>
>    1. when use uuid_kill of above uuid, FS reports ?No Such Channel?
>    2. then at 60000ms interval, FS will give messages like ?Originate
>    attempt x/3 in 60000ms?
>    3. when x reaches 1, the channel will be cleaned.
>
>
>
> On step 3 and 4, if there?s ?No Such Channel?, isn?t that the channel
> should not be displayed?
>
>
>
> On step 5, although the ?originate? is attempted ( which shouldn?t,
> because the call is already hangup on step 2 ), but neither caller nor the
> callee is ringed.
>
>
>
> Also, is there a way to cancel ?originate? ? eg send a cancel after
> invite.
>
>
>
> Regards
>
> Scott
>
> _______________________________________________
> 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
>
>


-- 
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  Mon May  4 06:16:09 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 May 2009 08:16:09 -0500
Subject: [Freeswitch-dev] originate creates a channel that can not be
	killed
In-Reply-To: <4462DF002246D44CB823FA4F985539941426D33C27@oa-exchange1.oa.com.au>
References: <4462DF002246D44CB823FA4F985539941426D33C27@oa-exchange1.oa.com.au>
Message-ID: <38C776E5-D1DC-4BE4-9320-02801FC1AAD5@freeswitch.org>

Please update to SVN trunk by doing a complete fresh check out...  
sounds like you might have some build skew.  If the problem persists  
please open a jira http://jira.freeswitch.org

http://wiki.freeswitch.org/wiki/Reporting_Bugs

/b


On May 3, 2009, at 11:00 PM, Scott Shen wrote:

> Also, is there a way to cancel ?originate? ? eg send a cancel after  
> invite.

Brian West
brian at freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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From msc at freeswitch.org  Mon May  4 13:50:36 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 May 2009 13:50:36 -0700
Subject: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH mod_opal Now Officially
	In Beta
Message-ID: <87f2f3b90905041350q5c896820o537c17e35abd690@mail.gmail.com>

The FreeSWITCH team would like everyone to know that the mod_opal module is
now officially in beta. Please read this article for more information:
http://www.freeswitch.org/node/179

Many thanks to Robert Jongbloed and Craig Southeren of the OPAL project for
their many years of support for open source VoIP.

-Michael S Collins
http://www.cluecon.com
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From bwillis at teldio.com  Tue May  5 11:51:41 2009
From: bwillis at teldio.com (Benjamin Willis)
Date: Tue, 5 May 2009 14:51:41 -0400
Subject: [Freeswitch-dev] FreeSWITCH Nortel Gateway Interop
Message-ID: <AC1EE595-C31A-4C50-AF40-3997E796BAFD@teldio.com>

Greetings,

We are performing interop testing to have FreeSWITCH as a gateway  
with a Nortel CS1000 box and would appreciate any help resolving our  
issues.  There is a box that is a registrar and a redirect server.   
We can successfully register against this box.  The box is picky  
about the invites we send; it expects something like the following :

INVITE sip:2270;phone-context=cdp.udp at newdomain.com;user=phone SIP/2.0

After great help from the freeSWITCH IRC we came up with the  
following dialplan entry to achieve this style header :

<action application="bridge" data="sofia/internal/sip:$1;phone- 
context=cdp.udp at newdomain.com;user=phone;fs_path=sip: 
192.168.15.8:5060;"/>

At this point we are sending the formatted INVITE message as the  
redirect server expects.  Our next issue where we would appreciate  
help is after the INVITE is sent, the redirect server sends back a  
302 Moved Temporarily as expected.

    SIP/2.0 302 Moved Temporarily
    Via: SIP/2.0/TCP  
192.168.15.11;branch=z9hG4bKy0NcZvFcNr1ZB;received=192.168.15.11
    From: "User 210" <sip:2300 at 192.168.15.11>;tag=jjmgrmHN6Qjaa
    To: <sip:2270;phone- 
context=cdp.udp at newdomain.com;user=phone>;tag=49582
    Call-ID: 6998551e-b04c-122c-2d80-39a48cb53b8d
    CSeq: 114437328 INVITE
    Contact: <sip:2270;phone-context=cdp.udp at newdomain.com: 
5060;maddr=192.168.15.10;transport=tcp;user=phone;x-nt- 
redirect=redirect-server>
    Content-Length: 0

The subsequent INVITE message is, again, sent back to the redirect  
server, not to the machine indicated by the maddr entry in the  
Contact field.  The new INVITE looks like this :

INVITE sip:2270;phone-context=cdp.udp at newdomain.com: 
5060;maddr=192.168.15.10;transport=tcp;user=phone;x-nt- 
redirect=redirect-server SIP/2.0
Via: SIP/2.0/UDP 192.168.15.11;rport;branch=z9hG4bK93BXjtmKD9pcN
Route: <sip:192.168.15.8:5060>
Max-Forwards: 69
From: "User 210" <sip:2300 at 192.168.15.11>;tag=ppSKZmm0rpS3e
To: <sip:2270;phone-context=cdp.udp at newdomain.com;user=phone>
Call-ID: ce59ac9e-b058-122c-2780-39a48cb53b8d
CSeq: 114439989 INVITE
Contact: <sip:mod_sofia at 192.168.15.11:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163
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-Type: application/sdp
Content-Disposition: session
Content-Length: 360
Remote-Party-ID: "User 210" <sip: 
2300 at 192.168.15.11>;screen=yes;privacy=off

v=0
o=FreeSWITCH 6002387420083056305 1437766807142321498 IN IP4  
192.168.15.11
s=FreeSWITCH
c=IN IP4 192.168.15.11
t=0 0
m=audio 28220 RTP/AVP 10 0 8 9 3 101 13
a=rtpmap:10 L16/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

The following is a wireshark capture of the repeated INVITE messages  
back to the same machine, it repeats 3 times after the initial invite :

Conv.| Time    | 192.168.15.11     | 192.168.15.8      |
0    |17.284   |         INVITE SDP ( 16-bit audio, stereo g711U  
g711A ...2 GSM teleph)          |SIP From: sip:2300 at 192.168.15.11  
To:sip:2270
      |         |(5060)   ------------------>  (5060)   |
0    |17.291   |         302 Moved Temporarily          |SIP Status
      |         |(5060)   <------------------  (5060)   |
0    |17.291   |         ACK       |                   |SIP Request
      |         |(5060)   ------------------>  (5060)   |
---------------------------------------------------------
1    |17.291   |         INVITE SDP ( 16-bit audio, stereo g711U  
g711A ...2 GSM teleph)          |SIP From: sip:2300 at 192.168.15.11  
To:sip:2270
      |         |(5060)   ------------------>  (5060)   |
1    |17.298   |         302 Moved Temporarily          |SIP Status
      |         |(5060)   <------------------  (5060)   |
1    |17.298   |         ACK       |                   |SIP Request
      |         |(5060)   ------------------>  (5060)   |
---------------------------------------------------------
2    |17.298   |         INVITE SDP ( 16-bit audio, stereo g711U  
g711A ...2 GSM teleph)          |SIP From: sip:2300 at 192.168.15.11  
To:sip:2270
      |         |(5060)   ------------------>  (5060)   |
2    |17.304   |         302 Moved Temporarily          |SIP Status
      |         |(5060)   <------------------  (5060)   |
2    |17.305   |         ACK       |                   |SIP Request
      |         |(5060)   ------------------>  (5060)   |
---------------------------------------------------------
3    |17.305   |         INVITE SDP ( 16-bit audio, stereo g711U  
g711A ...2 GSM teleph)          |SIP From: sip:2300 at 192.168.15.11  
To:sip:2270
      |         |(5060)   ------------------>  (5060)   |
3    |17.311   |         302 Moved Temporarily          |SIP Status
      |         |(5060)   <------------------  (5060)   |
3    |17.311   |         ACK       |                   |SIP Request
      |         |(5060)   ------------------>  (5060)   |

At this point we are a little stuck.  We feel that the use of fs_path  
may be having unwanted effects on our redirects after looking at  
sofia_glue.c:sofia_glue_do_invite() and the parsing of the fs_path  
variable.  We haven't filed this as a bug as it may be intended  
functionality and would like to get some insight from the more  
experience members before going down such a path.

Thanks in advance.



From msc at freeswitch.org  Tue May  5 17:48:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 May 2009 17:48:21 -0700
Subject: [Freeswitch-dev] ClueCon 2009 Blog, Moises Silva Speaking
Message-ID: <87f2f3b90905051748s788914c7x6fdd557be342f64a@mail.gmail.com>

FYI,

I just wanted to let the community know that we maintain a
blog<http://cluecon.com/blog/1>on the ClueCon
website <http://www.cluecon.com>. Please check it out. The latest entry
mentions Moises Silva's recent blog entry about his speaking at ClueCon this
year.

Please check the ClueCon blog periodically as we will be adding new
information about speakers, sponsors, and other good stuff.

-Michael S Collins
http://www.cluecon.com
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From sshen at oa.com.au  Tue May  5 18:52:40 2009
From: sshen at oa.com.au (Scott Shen)
Date: Wed, 6 May 2009 11:52:40 +1000
Subject: [Freeswitch-dev] originate creates a channel that can not be
 killed
In-Reply-To: <191c3a030905040611r68fc3973ufb77a8cf83d5740f@mail.gmail.com>
Message-ID: <4462DF002246D44CB823FA4F98553994144B14F992@oa-exchange1.oa.com.au>

Tried the latest check out from svn, the "un-killable" channel is not appear after step 2 this time. However, the retries is not working as expected,

For settings like {originate_timeout=6,originate_retries=3,originate_retry_sleep_ms=5000}, I would expect FS to ring the first party 3 times, each time ring for about 6 seconds and then sleep for 5 seconds, is that correct?

However, the first party was ringed 9 times, each with 6 seconds ringing and 5 seconds gap. In the console log, every 3 attempts, there's a

"switch_ivr_originate.c:1495 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [NO_ANSWER]"

Is there anything wrong with the retry counter? Or is there any conflict on the parameters I set?


Thanks & Regards
Scott



-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Monday, 4 May 2009 11:12 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] originate creates a channel that can not be killed

Please do not report bugs on the mailing list.  It is impossible to manage them from here.

The proper place to submit bug reports is

http://jira.freeswitch.org

On Sun, May 3, 2009 at 11:00 PM, Scott Shen <sshen at oa.com.au<mailto:sshen at oa.com.au>> wrote:

I am using FS r13096, found following problem


1.      originate {originate_timeout=6,originate_retries=3,originate_retry_sleep_ms=60000}user/1002 &bridge(user/1007)
2.      when 1002 side ringing, quickly pickup and hangup
3.      show channels gives something like

31b614d0-999d-406e-bdba-b18ead0d26d8,outbound,2009-05-04 13:42:56,1241408576,sofia/internal/sip:1002 at 192.168.10.100:5060;line=xStcbdok,CS_EXECUTE,FreeSWITCH,0000000000,,sip:1002 at 192.168.10.100:5060;line=xStcbdok,bridge,user/1007,,default,PCMA,8000,PCMA,8000
4.      when use uuid_kill of above uuid, FS reports "No Such Channel"
5.      then at 60000ms interval, FS will give messages like 'Originate attempt x/3 in 60000ms'
6.      when x reaches 1, the channel will be cleaned.



On step 3 and 4, if there's "No Such Channel", isn't that the channel should not be displayed?



On step 5, although the "originate" is attempted ( which shouldn't, because the call is already hangup on step 2 ), but neither caller nor the callee is ringed.



Also, is there a way to cancel "originate" ? eg send a cancel after invite.



Regards

Scott

_______________________________________________
Freeswitch-dev mailing list
Freeswitch-dev at lists.freeswitch.org<mailto: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



--
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
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From brian at freeswitch.org  Tue May  5 19:05:56 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 May 2009 21:05:56 -0500
Subject: [Freeswitch-dev] originate creates a channel that can not be
	killed
In-Reply-To: <4462DF002246D44CB823FA4F98553994144B14F992@oa-exchange1.oa.com.au>
References: <4462DF002246D44CB823FA4F98553994144B14F992@oa-exchange1.oa.com.au>
Message-ID: <034E80D9-7F69-43B1-BCF4-881DD67F6C68@freeswitch.org>

Can you give a more detailed step by step and place it on jira.

/b

On May 5, 2009, at 8:52 PM, Scott Shen wrote:

> Tried the latest check out from svn, the ?un-killable? channel is  
> not appear after step 2 this time. However, the retries is not  
> working as expected,
>
> For settings like  
> {originate_timeout 
> =6,originate_retries=3,originate_retry_sleep_ms=5000}, I would  
> expect FS to ring the first party 3 times, each time ring for about  
> 6 seconds and then sleep for 5 seconds, is that correct?
>
> However, the first party was ringed 9 times, each with 6 seconds  
> ringing and 5 seconds gap. In the console log, every 3 attempts,  
> there?s a
>
> ?switch_ivr_originate.c:1495 switch_ivr_originate() Cannot create  
> outgoing channel of type [user] cause: [NO_ANSWER]?
>
> Is there anything wrong with the retry counter? Or is there any  
> conflict on the parameters I set?
>
>
> Thanks & Regards
> Scott
>

Brian West
brian at freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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From anthony.minessale at gmail.com  Tue May  5 19:46:25 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 May 2009 21:46:25 -0500
Subject: [Freeswitch-dev] originate creates a channel that can not be
	killed
In-Reply-To: <034E80D9-7F69-43B1-BCF4-881DD67F6C68@freeswitch.org>
References: <4462DF002246D44CB823FA4F98553994144B14F992@oa-exchange1.oa.com.au>
	<034E80D9-7F69-43B1-BCF4-881DD67F6C68@freeswitch.org>
Message-ID: <191c3a030905051946k362e93a2r136dfe819ea83adb@mail.gmail.com>

originate_timeout is global to the whole originate process
you need leg_timeout=6

On Tue, May 5, 2009 at 9:05 PM, Brian West <brian at freeswitch.org> wrote:

> Can you give a more detailed step by step and place it on jira.
> /b
>
> On May 5, 2009, at 8:52 PM, Scott Shen wrote:
>
> Tried the latest check out from svn, the ?un-killable? channel is not
> appear after step 2 this time. However, the retries is not working as
> expected,
>
> For settings like {originate_timeout=6,originate_retries=3,originate_retry_sleep_ms=5000},
> I would expect FS to ring the first party 3 times, each time ring for about
> 6 seconds and then sleep for 5 seconds, is that correct?
>
> However, the first party was ringed 9 times, each with 6 seconds ringing
> and 5 seconds gap. In the console log, every 3 attempts, there?s a
>
> ?switch_ivr_originate.c:1495 switch_ivr_originate() Cannot create outgoing
> channel of type [user] cause: [NO_ANSWER]?
>
> Is there anything wrong with the retry counter? Or is there any conflict on
> the parameters I set?
>
>
> Thanks & Regards
> Scott
>
>
>
> Brian West
> brian at freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
> _______________________________________________
> 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
>
>


-- 
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 sshen at oa.com.au  Tue May  5 20:21:28 2009
From: sshen at oa.com.au (Scott Shen)
Date: Wed, 6 May 2009 13:21:28 +1000
Subject: [Freeswitch-dev] originate creates a channel that can not
	be	killed
In-Reply-To: <191c3a030905051946k362e93a2r136dfe819ea83adb@mail.gmail.com>
Message-ID: <4462DF002246D44CB823FA4F98553994144B14F993@oa-exchange1.oa.com.au>

Same result.

Raised a jira FSCORE-361<http://jira.freeswitch.org/browse/FSCORE-361>



-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Wednesday, 6 May 2009 12:46 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] originate creates a channel that can not be killed

originate_timeout is global to the whole originate process
you need leg_timeout=6
On Tue, May 5, 2009 at 9:05 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Can you give a more detailed step by step and place it on jira.

/b

On May 5, 2009, at 8:52 PM, Scott Shen wrote:


Tried the latest check out from svn, the "un-killable" channel is not appear after step 2 this time. However, the retries is not working as expected,

For settings like {originate_timeout=6,originate_retries=3,originate_retry_sleep_ms=5000}, I would expect FS to ring the first party 3 times, each time ring for about 6 seconds and then sleep for 5 seconds, is that correct?

However, the first party was ringed 9 times, each with 6 seconds ringing and 5 seconds gap. In the console log, every 3 attempts, there's a

"switch_ivr_originate.c:1495 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [NO_ANSWER]"

Is there anything wrong with the retry counter? Or is there any conflict on the parameters I set?


Thanks & Regards
Scott


Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

-- Meet us at ClueCon!  http://www.cluecon.com<http://www.cluecon.com/>





_______________________________________________
Freeswitch-dev mailing list
Freeswitch-dev at lists.freeswitch.org<mailto:Freeswitch-dev at lists.freeswitch.org>
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://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
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From sshen at oa.com.au  Tue May  5 22:44:24 2009
From: sshen at oa.com.au (Scott Shen)
Date: Wed, 6 May 2009 15:44:24 +1000
Subject: [Freeswitch-dev] fifo in occasionally drops call
Message-ID: <4462DF002246D44CB823FA4F98553994144B14F994@oa-exchange1.oa.com.au>

Using FS r13231,  when testing fifo in, experienced occasionally call drops with no obvious reason, is it something to do with early media? See the debug log below


Dialplan: sofia/internal/sip:1007 at 192.168.0.31:5342 Regex (PASS) [TEST_Q] destination_number(40001) =~ /^40001$/ break=on-false
Dialplan: sofia/internal/sip:1007 at 192.168.0.31:5342 Action fifo(test_q in)
2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/sip:1007 at 192.168.0.31:5342) State Change CS_ROUTING -> CS_EXECUTE
2009-05-06 15:18:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1007 at 192.168.0.31:5342 [BREAK]
2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/sip:1007 at 192.168.0.31:5342) State ROUTING going to sleep
2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/sip:1007 at 192.168.0.31:5342) Running State Change CS_EXECUTE
2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/sip:1007 at 192.168.0.31:5342) State EXECUTE
2009-05-06 15:18:46 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/sip:1007 at 192.168.0.31:5342 SOFIA EXECUTE
2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/sip:1007 at 192.168.0.31:5342 Standard EXECUTE
EXECUTE sofia/internal/sip:1007 at 192.168.0.31:5342 set(outside_call=true)
2009-05-06 15:18:46 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/sip:1007 at 192.168.0.31:5342 SET [outside_call]=[true]
EXECUTE sofia/internal/sip:1007 at 192.168.0.31:5342 fifo(test_q in)
2009-05-06 15:18:46 [DEBUG] switch_ivr.c:1342 switch_ivr_session_transfer() (sofia/internal/sip:1007 at 192.168.0.31:5342) State Change CS_EXECUTE -> CS_ROUTING
2009-05-06 15:18:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1007 at 192.168.0.31:5342 [BREAK]
2009-05-06 15:18:46 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/sip:1007 at 192.168.0.31:5342 [CS_EXECUTE] [NORMAL_CLEARING]
2009-05-06 15:18:46 [DEBUG] switch_channel.c:1641 switch_channel_perform_hangup() Send signal sofia/internal/sip:1007 at 192.168.0.31:5342 [KILL]
2009-05-06 15:18:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1007 at 192.168.0.31:5342 [BREAK]
2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/sip:1007 at 192.168.0.31:5342) State EXECUTE going to sleep
2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/sip:1007 at 192.168.0.31:5342) Running State Change CS_HANGUP
2009-05-06 15:18:46 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/sip:1007 at 192.168.0.31:5342) State HANGUP
2009-05-06 15:18:46 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/internal/sip:1007 at 192.168.0.31:5342 hanging up, cause: NORMAL_CLEARING
2009-05-06 15:18:46 [DEBUG] mod_sofia.c:378 sofia_on_hangup() Sending BYE to sofia/internal/sip:1007 at 192.168.0.31:5342


Regards
Scott
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From msc at freeswitch.org  Wed May  6 07:06:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 May 2009 07:06:03 -0700
Subject: [Freeswitch-dev] ANNOUNCEMENT: FreeSWITCH 1.0.4pre7 Now Available
Message-ID: <87f2f3b90905060706j7c8d4d2dh5bd620db65ab52a@mail.gmail.com>

FYI,

Please update your installations as soon as possible. More information on
this update is available here <http://www.freeswitch.org/node/181>.

Thanks for all of your feedback - please keep it coming and join us on IRC
if you have any questions about the newest version.

-Michael S Collins
http://www.cluecon.com
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From msc at freeswitch.org  Wed May  6 09:36:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 May 2009 09:36:34 -0700
Subject: [Freeswitch-dev] Interesting Blog About HD Telephony
Message-ID: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com>

FYI,

Here's a nice story <http://www.freeswitch.org/node/182> for you all to
check out. Please check it out and pass it on.

-Michael
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From gmaruzz at celliax.org  Wed May  6 10:00:57 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 6 May 2009 19:00:57 +0200
Subject: [Freeswitch-dev] Interesting Blog About HD Telephony
In-Reply-To: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com>
References: <87f2f3b90905060936o5655b2c9g7a5a339ce7411518@mail.gmail.com>
Message-ID: <7b197bef0905061000v56269cceh73c8b7519662275e@mail.gmail.com>

Ciao Michael,
if you like, you can add that using mod_skypiax you have native hd
skype->FS and FS->Skype (no hardware needed) :-)



On Wed, May 6, 2009 at 6:36 PM, Michael Collins <msc at freeswitch.org> wrote:
> FYI,
>
> Here's a nice story for you all to check out. Please check it out and pass
> it on.
>
> -Michael
>
> _______________________________________________
> 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 raison at chatsubo.net  Fri May  8 10:22:47 2009
From: raison at chatsubo.net (Kevin Raison)
Date: Fri, 08 May 2009 10:22:47 -0700
Subject: [Freeswitch-dev] Adding a new embedded language
Message-ID: <4A046A67.7070007@chatsubo.net>

I am thinking about writing a new embedded language module for 
Freeswitch using Common Lisp, specifically ECL 
(http://ecls.sourceforge.net/).  Beyond looking over the code in the 
Freeswitch repository for the other embedded languages, does anyone have 
a recommendation on how to begin?  I don't see any documentation 
specific to this task;  does it exist somewhere?

Thanks in advance for your help.
Kevin Raison


From anthony.minessale at gmail.com  Fri May  8 11:10:01 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 May 2009 13:10:01 -0500
Subject: [Freeswitch-dev] Adding a new embedded language
In-Reply-To: <4A046A67.7070007@chatsubo.net>
References: <4A046A67.7070007@chatsubo.net>
Message-ID: <191c3a030905081110t7aaf9bdcp2fbde9e79c818dc3@mail.gmail.com>

all the languages besides javascript use swig to tie the language to FS
mod_spidermonkey is done by hand.

Essentially, there is a switch_cpp.cpp in the core that is ideal for
embedding by abstracting the objects

if you look at ESL, there is a similar approach to external control.


On Fri, May 8, 2009 at 12:22 PM, Kevin Raison <raison at chatsubo.net> wrote:

> I am thinking about writing a new embedded language module for
> Freeswitch using Common Lisp, specifically ECL
> (http://ecls.sourceforge.net/).  Beyond looking over the code in the
> Freeswitch repository for the other embedded languages, does anyone have
> a recommendation on how to begin?  I don't see any documentation
> specific to this task;  does it exist somewhere?
>
> Thanks in advance for your help.
> Kevin Raison
>
> _______________________________________________
> 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
>



-- 
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 bwillis at teldio.com  Fri May  8 19:44:57 2009
From: bwillis at teldio.com (Benjamin Willis)
Date: Fri, 8 May 2009 22:44:57 -0400
Subject: [Freeswitch-dev] FreeSWITCH Nortel Gateway Interop
In-Reply-To: <AC1EE595-C31A-4C50-AF40-3997E796BAFD@teldio.com>
References: <AC1EE595-C31A-4C50-AF40-3997E796BAFD@teldio.com>
Message-ID: <08AB8A7D-7B2A-4403-BFF3-37F9A8094833@teldio.com>

I guess the issue was a little too in-depth for assistance.  Anyhow,  
we have resolved this issue by using maddr instead of fs_path and now  
have calls working between a FreeSWITCH and Nortel system which we  
have started to document http://wiki.freeswitch.org/wiki/ 
Connecting_Freeswitch_And_Nortel.

Ben

On May 5, 2009, at 2:51 PM, Benjamin Willis wrote:

> Greetings,
>
> We are performing interop testing to have FreeSWITCH as a gateway
> with a Nortel CS1000 box and would appreciate any help resolving our
> issues.  There is a box that is a registrar and a redirect server.
> We can successfully register against this box.  The box is picky
> about the invites we send; it expects something like the following :
>
> INVITE sip:2270;phone-context=cdp.udp at newdomain.com;user=phone SIP/2.0
>
> After great help from the freeSWITCH IRC we came up with the
> following dialplan entry to achieve this style header :
>
> <action application="bridge" data="sofia/internal/sip:$1;phone-
> context=cdp.udp at newdomain.com;user=phone;fs_path=sip:
> 192.168.15.8:5060;"/>
>
> At this point we are sending the formatted INVITE message as the
> redirect server expects.  Our next issue where we would appreciate
> help is after the INVITE is sent, the redirect server sends back a
> 302 Moved Temporarily as expected.
>
>     SIP/2.0 302 Moved Temporarily
>     Via: SIP/2.0/TCP
> 192.168.15.11;branch=z9hG4bKy0NcZvFcNr1ZB;received=192.168.15.11
>     From: "User 210" <sip:2300 at 192.168.15.11>;tag=jjmgrmHN6Qjaa
>     To: <sip:2270;phone-
> context=cdp.udp at newdomain.com;user=phone>;tag=49582
>     Call-ID: 6998551e-b04c-122c-2d80-39a48cb53b8d
>     CSeq: 114437328 INVITE
>     Contact: <sip:2270;phone-context=cdp.udp at newdomain.com:
> 5060;maddr=192.168.15.10;transport=tcp;user=phone;x-nt-
> redirect=redirect-server>
>     Content-Length: 0
>
> The subsequent INVITE message is, again, sent back to the redirect
> server, not to the machine indicated by the maddr entry in the
> Contact field.  The new INVITE looks like this :
>
> INVITE sip:2270;phone-context=cdp.udp at newdomain.com:
> 5060;maddr=192.168.15.10;transport=tcp;user=phone;x-nt-
> redirect=redirect-server SIP/2.0
> Via: SIP/2.0/UDP 192.168.15.11;rport;branch=z9hG4bK93BXjtmKD9pcN
> Route: <sip:192.168.15.8:5060>
> Max-Forwards: 69
> From: "User 210" <sip:2300 at 192.168.15.11>;tag=ppSKZmm0rpS3e
> To: <sip:2270;phone-context=cdp.udp at newdomain.com;user=phone>
> Call-ID: ce59ac9e-b058-122c-2780-39a48cb53b8d
> CSeq: 114439989 INVITE
> Contact: <sip:mod_sofia at 192.168.15.11:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163
> 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-Type: application/sdp
> Content-Disposition: session
> Content-Length: 360
> Remote-Party-ID: "User 210" <sip:
> 2300 at 192.168.15.11>;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 6002387420083056305 1437766807142321498 IN IP4
> 192.168.15.11
> s=FreeSWITCH
> c=IN IP4 192.168.15.11
> t=0 0
> m=audio 28220 RTP/AVP 10 0 8 9 3 101 13
> a=rtpmap:10 L16/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> The following is a wireshark capture of the repeated INVITE messages
> back to the same machine, it repeats 3 times after the initial  
> invite :
>
> Conv.| Time    | 192.168.15.11     | 192.168.15.8      |
> 0    |17.284   |         INVITE SDP ( 16-bit audio, stereo g711U
> g711A ...2 GSM teleph)          |SIP From: sip:2300 at 192.168.15.11
> To:sip:2270
>       |         |(5060)   ------------------>  (5060)   |
> 0    |17.291   |         302 Moved Temporarily          |SIP Status
>       |         |(5060)   <------------------  (5060)   |
> 0    |17.291   |         ACK       |                   |SIP Request
>       |         |(5060)   ------------------>  (5060)   |
> ---------------------------------------------------------
> 1    |17.291   |         INVITE SDP ( 16-bit audio, stereo g711U
> g711A ...2 GSM teleph)          |SIP From: sip:2300 at 192.168.15.11
> To:sip:2270
>       |         |(5060)   ------------------>  (5060)   |
> 1    |17.298   |         302 Moved Temporarily          |SIP Status
>       |         |(5060)   <------------------  (5060)   |
> 1    |17.298   |         ACK       |                   |SIP Request
>       |         |(5060)   ------------------>  (5060)   |
> ---------------------------------------------------------
> 2    |17.298   |         INVITE SDP ( 16-bit audio, stereo g711U
> g711A ...2 GSM teleph)          |SIP From: sip:2300 at 192.168.15.11
> To:sip:2270
>       |         |(5060)   ------------------>  (5060)   |
> 2    |17.304   |         302 Moved Temporarily          |SIP Status
>       |         |(5060)   <------------------  (5060)   |
> 2    |17.305   |         ACK       |                   |SIP Request
>       |         |(5060)   ------------------>  (5060)   |
> ---------------------------------------------------------
> 3    |17.305   |         INVITE SDP ( 16-bit audio, stereo g711U
> g711A ...2 GSM teleph)          |SIP From: sip:2300 at 192.168.15.11
> To:sip:2270
>       |         |(5060)   ------------------>  (5060)   |
> 3    |17.311   |         302 Moved Temporarily          |SIP Status
>       |         |(5060)   <------------------  (5060)   |
> 3    |17.311   |         ACK       |                   |SIP Request
>       |         |(5060)   ------------------>  (5060)   |
>
> At this point we are a little stuck.  We feel that the use of fs_path
> may be having unwanted effects on our redirects after looking at
> sofia_glue.c:sofia_glue_do_invite() and the parsing of the fs_path
> variable.  We haven't filed this as a bug as it may be intended
> functionality and would like to get some insight from the more
> experience members before going down such a path.
>
> Thanks in advance.
>
>
> _______________________________________________
> 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 tzury.by at gmail.com  Fri May  8 13:38:53 2009
From: tzury.by at gmail.com (Tzury Bar Yochay)
Date: Fri, 8 May 2009 23:38:53 +0300
Subject: [Freeswitch-dev] Seeking for a developer for a small application on
	top of LibSangoma
In-Reply-To: <471deb400905081337t7b5a4f01ob5c10041732b7a7b@mail.gmail.com>
References: <471deb400905081337t7b5a4f01ob5c10041732b7a7b@mail.gmail.com>
Message-ID: <471deb400905081338x340b1f53l5b19b75c34d93876@mail.gmail.com>

 Hi,

I would like to build a server which receives modem calls (V.32/V.110) on
one end, and passes the data received to an IP server on another end. It
then, waits for a reply from the server and pass the reply back to the
endpoint device over the modem session.I put a description of the project
at: http://www.reguluslabs.com/pub/sangoma-e1-ip-bridge.
The project itself consists of a full blown socket application which we
intend to write by ourselves.
The help we need is to design and write a wrapper around LibSangoma in order
to ease our development.
A draft of the wrapper is located at :
http://www.reguluslabs.com/pub/sangoma-e1-ip-bridge-interface.

If you have an experience with the Sangoma API and think you can join forces
with us developing this during the coming weeks please reply to me.

Looking forward hearing from you,
Tzury Bar Yochay
Regulus Labs ltd.

tzury.by at reguluslabs.com
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From anthony.minessale at gmail.com  Mon May 11 10:31:45 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 May 2009 12:31:45 -0500
Subject: [Freeswitch-dev] Seeking for a developer for a small
	application on top of LibSangoma
In-Reply-To: <471deb400905081338x340b1f53l5b19b75c34d93876@mail.gmail.com>
References: <471deb400905081337t7b5a4f01ob5c10041732b7a7b@mail.gmail.com>
	<471deb400905081338x340b1f53l5b19b75c34d93876@mail.gmail.com>
Message-ID: <191c3a030905111031uf9c3133y3c10cefce8849d47@mail.gmail.com>

you may find this already exists using the sangoma driver for OpenZAP
bundled with FreeSWITCH


On Fri, May 8, 2009 at 3:38 PM, Tzury Bar Yochay <tzury.by at gmail.com> wrote:

> Hi,
>
> I would like to build a server which receives modem calls (V.32/V.110) on
> one end, and passes the data received to an IP server on another end. It
> then, waits for a reply from the server and pass the reply back to the
> endpoint device over the modem session.I put a description of the project
> at: http://www.reguluslabs.com/pub/sangoma-e1-ip-bridge.
> The project itself consists of a full blown socket application which we
> intend to write by ourselves.
> The help we need is to design and write a wrapper around LibSangoma in
> order to ease our development.
> A draft of the wrapper is located at :
> http://www.reguluslabs.com/pub/sangoma-e1-ip-bridge-interface.
>
> If you have an experience with the Sangoma API and think you can join
> forces with us developing this during the coming weeks please reply to me.
>
> Looking forward hearing from you,
> Tzury Bar Yochay
> Regulus Labs ltd.
>
> tzury.by at reguluslabs.com
>
>
> _______________________________________________
> 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
>
>


-- 
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 May 11 10:34:20 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 May 2009 10:34:20 -0700
Subject: [Freeswitch-dev] Cluecon 2009 News
Message-ID: <87f2f3b90905111034g71497662m9b2e623862d2862d@mail.gmail.com>

FYI, for those of you keeping up on ClueCon 2009 please visit the latest
blog entry:
http://cluecon.com/node/29

Thanks,
Michael
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From msc at freeswitch.org  Tue May 12 11:00:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 May 2009 11:00:37 -0700
Subject: [Freeswitch-dev] ANNOUNCEMENT: www.freeswitch.org site updated
Message-ID: <87f2f3b90905121100m4346bb76q7e12f2e415cf0908@mail.gmail.com>

FYI,

We'd like to let the community know that we've moved the main site,
www.freeswitch.org, to a new server and have updated it to the latest Drupal
version. We are still calibrating the server so if the site seems slower
than usual please let me know off list and we'll take a look. Also, because
this upgrade was so major we were forced to wipe out all the users from the
previous Drupal install. Please feel free to sign up again if you so desire.
You only need to sign up if you wish to post comments.

One other new feature is the "Share This" button. (You need JavaScript
enabled to see this.) It makes it as easy as possible to share stories with
Digg, SU, Deli.ico.us, etc. If there are any stories you've not already dugg
then please try out the Share This button to see how easy it is to digg
stuff now.

As always, if you have any questions or comments then please email me off
list and I'll address them quickly.

-Michael
http://www.cluecon.com
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From msc at freeswitch.org  Tue May 12 11:24:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 May 2009 11:24:37 -0700
Subject: [Freeswitch-dev] Cluecon 2009 - Hotel Special!
Message-ID: <87f2f3b90905121124s7e4c8d22ye3c6d287834acfd4@mail.gmail.com>

Good news! For those of you who like to get a good deal on a hotel room
we've got the inside scoop on a special for the Wyndham Chicago. I just did
a search on expedia.com for a hotel with these specs:
Hotel only
Check-in: 8/3/2009, Check-Out: 8/6/2009
1 room, 2 adults
City: Chicago
Hotel Name: "Wyndham Chicago"

The results were very nice: the non-refundable rate per night is only
$140!!! Get to expedia.com NOW and get signed up!

We look forward to seeing you in Chicago this August.
-Michael
http://www.cluecon.com
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From gmaruzz at celliax.org  Tue May 12 12:24:17 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 12 May 2009 21:24:17 +0200
Subject: [Freeswitch-dev] [Freeswitch-users] Cluecon 2009 - Hotel
	Special!
In-Reply-To: <87f2f3b90905121124s7e4c8d22ye3c6d287834acfd4@mail.gmail.com>
References: <87f2f3b90905121124s7e4c8d22ye3c6d287834acfd4@mail.gmail.com>
Message-ID: <7b197bef0905121224u126caa5lb88ea5a56eef45f6@mail.gmail.com>

yay Michael! Got it! I owe you one!!!!!



On Tue, May 12, 2009 at 8:24 PM, Michael Collins <msc at freeswitch.org> wrote:
> Good news! For those of you who like to get a good deal on a hotel room
> we've got the inside scoop on a special for the Wyndham Chicago. I just did
> a search on expedia.com for a hotel with these specs:
> Hotel only
> Check-in: 8/3/2009, Check-Out: 8/6/2009
> 1 room, 2 adults
> City: Chicago
> Hotel Name: "Wyndham Chicago"
>
> The results were very nice: the non-refundable rate per night is only
> $140!!! Get to expedia.com NOW and get signed up!
>
> We look forward to seeing you in Chicago this August.
> -Michael
> http://www.cluecon.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 marketing at cluecon.com  Thu May 14 13:14:41 2009
From: marketing at cluecon.com (Michael Collins)
Date: Thu, 14 May 2009 13:14:41 -0700
Subject: [Freeswitch-dev] Call For Participants: Lightning Talks at ClueCon
	2009
Message-ID: <87f2f3b90905141314w3a2b24ccu41c692588555264b@mail.gmail.com>

*ClueCon 2009 is coming soon!*

We are interested in your thoughts on subjects for lighting talks. We would
love to have a number of 5-10 minute presentations by members of the
community. If you would like to give a talk, or just have an idea for a
talk, please let us know.

How do lightning talks work? Quite simply, the presenter has just a few
minutes to speak on a particular subject, usually no more than 10 minutes.
He or she will deliver the information rapidly, which means keeping the
presentation focused tightly on the subject being discussed. Lightning talks
usually do not have enough time for audience Q&A. However, ClueCon has a
long lunch period that is designed to allow attendees plenty of time to
interact. Those are perfect times to discuss lightning talks or any other
presentations. Those who give presentations enjoy interacting with other
attendees in a relaxed atmosphere during lunch or in the evening at dinner.

If you haven't already registered for ClueCon 2009 then please call us at
877.742.CLUE right away and we will complete your registration. Also, don't
forget that expedia.com has some nice hotel deals for the Wyndham Chicago.
Book your room today!

We look forward to hearing from you and seeing you all at ClueCon in
Chicago.
-Michael
http://www.cluecon.com
877.742.CLUE
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From regs at kinetix.gr  Mon May 18 05:00:23 2009
From: regs at kinetix.gr (Apostolos Pantsiopoulos)
Date: Mon, 18 May 2009 15:00:23 +0300
Subject: [Freeswitch-dev] mod_opal testing
Message-ID: <4A114DD7.8070309@kinetix.gr>

Hi,

While testing mod_opal I get the following message :

2009-05-18 11:57:04 [INFO] h323.cxx:4127 H323() CreateLogicalChannel - 
unknown data type

no matter what codec I use (even when the call is successful).


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
-------------------------------------------


From msc at freeswitch.org  Tue May 19 12:48:18 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 May 2009 12:48:18 -0700
Subject: [Freeswitch-dev] New ZDNet Article About FreeSWITCH
Message-ID: <87f2f3b90905191248u2813e829veb4e7455b7de535e@mail.gmail.com>

Gang,

Please visit the main FreeSWITCH site and check out the
links<http://www.freeswitch.org/node/185>to the new ZDNet story.
Please spread the word with all of the link sharing
sites that you have.

Thanks!
Michael S Collins

http://www.cluecon.com
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From solko at gcdf.pl  Thu May 21 08:00:55 2009
From: solko at gcdf.pl (Szymon Olko)
Date: Thu, 21 May 2009 17:00:55 +0200
Subject: [Freeswitch-dev] ignore_early_media and mod_conference two
	different cases
Message-ID: <4A156CA7.7010108@gcdf.pl>

Hi,

I made scenario in which I originate call and put it to conference. I use events to check if call was answered or not. I do it all
with "ignore_early_media" set to false, so I can record/hear what remote leg is saying, this way call center operators can help to
teach my system to recognize it automatically for next calls. I wrote that having one member in conference that I put this call
in. I use that command.

ORIGINATE {camel_command_id=4}sofia/gateway/sip.freeconet.pl/664558242 &conference(digitalvoice_call#0 at talk) '' ''  ULTIMO 429955

Call is not answered, it is rejected on phone. There is one member in conference.

I get following events scenario (I omit here not interesting events for me):
CHANNEL_CREATE
CHANNEL_OUTGOING
CHANNEL_ORIGINATE
CHANNEL_STATE
CHANNEL_STATE
CHANNEL_STATE
CHANNEL_PROGRESS_MEDIA
CHANNEL_STATE
CHANNEL_EXECUTE
CUSTOM_conference::maintenance
CHANNEL_PROGRESS
CUSTOM_conference::maintenance
CHANNEL_EXECUTE_COMPLETE
CHANNEL_HANGUP
CHANNEL_DESTROY


If I do all the same but conference is not existing (there is no member in that conference), I get following events in order:
CHANNEL_CREATE
CHANNEL_OUTGOING
CHANNEL_ORIGINATE
CHANNEL_STATE
CHANNEL_STATE
CHANNEL_STATE
CHANNEL_PROGRESS_MEDIA
CHANNEL_STATE
CHANNEL_EXECUTE
CUSTOM_conference::maintenance
CHANNEL_ANSWER
CHANNEL_PROGRESS
CUSTOM_conference::maintenance
CHANNEL_EXECUTE_COMPLETE
CHANNEL_HANGUP
CHANNEL_DESTROY

Call was not answer so there should be no CHANNEL_ANSWER event but it is there.

In other cases when I answer call I get events order:
- when there is member in conference
 CHANNEL_PROGRESS before CHANNEL_ANSWER event, time between them is the time of phone ringing.
- where there is no member in conference
 CHANNEL_PROGRESS event 4 seconds after CHANNEL_ANSWER event,

I cannot use this what I made for empty conferences. Maybe I don't understand early_media, but I would like it to work like in NOT
 EMPTY conference case.
Is there a way to make some 'virtual/local' member that will be in conference just to not be empty?

Regards
Szymon


From msc at freeswitch.org  Thu May 21 20:54:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 May 2009 20:54:24 -0700
Subject: [Freeswitch-dev] IMPORTANT: Latest FreeSWITCH SVN Updates -
	Bootstrap Required
Message-ID: <C1BDA36C-CE12-4A98-AD57-EF073960D1AE@freeswitch.org>

FYI,

We just want to let everyone know that we have made a few updates that  
will require a rebootstrap. One of the key updates was a security fix  
for libsndfile. In this particular case it won't be possible simply to  
"make current" like you normally do. Here is a common set of commands  
for a typical Linux rebootstrap:

cd /usr/src/freeswitch.trunk
make clean
svn up
./bootstrap.sh
./configure
make install

NOTE: if you've got the libzrtp file and you've already run the  
buildzrtp.sh script then be sure to use "./configure --enable-zrtp" in  
the above operation.

Thank you for your continued support of the FreeSWITCH project!

-Michael S Collins
http://www.cluecon.com
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From dftoro at yahoo.com  Fri May 22 06:09:49 2009
From: dftoro at yahoo.com (Diego Toro)
Date: Fri, 22 May 2009 06:09:49 -0700 (PDT)
Subject: [Freeswitch-dev] double re-register problem (machine avaible)
Message-ID: <309320.11806.qm@web33506.mail.mud.yahoo.com>

Greetings
?
I have the same problem reported in http://jira.freeswitch.org/browse/SFSIP-143.
I am running FS (rev 13424) on Windows and I trying register FS on Siemens HighPath 3000.
?
I have avaible on public IP the Siemens 3000 to debug, to reproduce and may be fix issue SFSIP-143 still open. 
?
I am begining with FreeSWITCH and i don't have skills yet to find the problem.

More information, send me a email

Thank you
?
Diego


      
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From brian at freeswitch.org  Fri May 22 07:14:04 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 May 2009 09:14:04 -0500
Subject: [Freeswitch-dev] double re-register problem (machine avaible)
In-Reply-To: <309320.11806.qm@web33506.mail.mud.yahoo.com>
References: <309320.11806.qm@web33506.mail.mud.yahoo.com>
Message-ID: <BDF96A50-980E-43AB-94AA-9B99A87C7868@freeswitch.org>

Are you 100% sure you have updated since I fixed the issue?  You'll  
have to download and compile it again.

/b

On May 22, 2009, at 8:09 AM, Diego Toro wrote:

> Greetings
>
> I have the same problem reported in http://jira.freeswitch.org/browse/SFSIP-143 
> .
> I am running FS (rev 13424) on Windows and I trying register FS on  
> Siemens HighPath 3000.
>
> I have avaible on public IP the Siemens 3000 to debug, to reproduce  
> and may be fix issue SFSIP-143 still open.
>
> I am begining with FreeSWITCH and i don't have skills yet to find  
> the problem.
>
> More information, send me a email
>
> Thank you
>
> Diego
>
> _____________

Brian West
brian at freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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From dftoro at yahoo.com  Fri May 22 14:23:08 2009
From: dftoro at yahoo.com (Diego Toro)
Date: Fri, 22 May 2009 14:23:08 -0700 (PDT)
Subject: [Freeswitch-dev] double re-register problem (machine avaible)
Message-ID: <126266.104.qm@web33501.mail.mud.yahoo.com>

Brian thank you? for your help.
?
I am testing again the integration between FS and Siemens pbx, I see now that setting expire-seconds on xml gateway? to a? value minor that 120 FS doen't lost the register (if this value is more than 120 the register is lost), but the calls are dropped after few seconds.
?
Diego

--- On Fri, 5/22/09, Brian West <brian at freeswitch.org> wrote:


From: Brian West <brian at freeswitch.org>
Subject: Re: [Freeswitch-dev] double re-register problem (machine avaible)
To: freeswitch-dev at lists.freeswitch.org
Date: Friday, May 22, 2009, 9:14 AM


Are you 100% sure you have updated since I fixed the issue? ?You'll have to download and compile it again.


/b



On May 22, 2009, at 8:09 AM, Diego Toro wrote:






Greetings
?
I have the same problem reported in?http://jira.freeswitch.org/browse/SFSIP-143.
I am running FS (rev 13424) on Windows and I trying register FS on Siemens HighPath 3000.
?
I have avaible on public IP the Siemens 3000 to debug, to reproduce and may be fix issue SFSIP-143 still open.
?
I am begining with FreeSWITCH and i don't have skills yet to find the problem.

More information, send me a email

Thank you
?
Diego
_____________





Brian West
brian at freeswitch.org

-- Meet us at ClueCon! ?http://www.cluecon.com






-----Inline Attachment Follows-----


_______________________________________________
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



      
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From brian at freeswitch.org  Tue May 26 09:36:23 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 26 May 2009 11:36:23 -0500
Subject: [Freeswitch-dev] Sounds order
Message-ID: <E1DB3D96-92EB-4763-9C1B-9AA1127836F9@freeswitch.org>

I'm getting ready for the next sound file order for FreeSWITCH.  I  
have a rather large set of files to be recorded for the zRTP  
integration if anyone wants to help out.  ;)  Please contact me off  
list.  I would like everyone to update and try out voicemail and  
nitpick anything that you feel is wrong there too and let me know so I  
can have them corrected in this order also.

Thanks,

Brian West
brian at freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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From brian at freeswitch.org  Tue May 26 10:34:48 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 26 May 2009 12:34:48 -0500
Subject: [Freeswitch-dev] STUN and rport on Polycom phones
Message-ID: <74FB7C03-EC28-4935-9C13-F25652704343@freeswitch.org>

It has come to my attention that Polycom hasn't had a business case to  
support rport and STUN.  If you can please kindly email Marek.Dutkiewicz at polycom.com 
  and let him know you would like to see STUN and rport support in the  
polycom products.  Its really one of the last things missing in the  
phone to make it easy to deploy in a NAT env.

Thanks,

Brian West
brian at freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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From brian at freeswitch.org  Tue May 26 12:47:52 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 26 May 2009 14:47:52 -0500
Subject: [Freeswitch-dev] Pre8 Release on Digg
Message-ID: <B193417D-6958-496A-9FD0-59CC131A8470@freeswitch.org>

Dear FreeSWITCHers,
	Now I'm gonna take a moment here to guilt each and everyone of you  
into checking out the story about Pre8 on Digg.  We have all worked  
long and hard to get to 1.0.4 and we still have a little bit to go.   
So everyone out there that asks "What can I do to help the project?",  
this is your chance to do so.  Help us to promote the project, which  
brings more people to help in supporting the community, the project  
and you the end user.

Also looking for people to help manage jira, test bugs, ask the right  
questions and line up the bugs so we can knock them out! Please email  
me if you're interested.

Here is the link for you to help out http://digg.com/search?s=FreeSWITCH+Pre8

Brian West
brian at freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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From echu at vseinc.com  Tue May 26 13:58:28 2009
From: echu at vseinc.com (echu at vseinc.com)
Date: Tue, 26 May 2009 16:58:28 -0400
Subject: [Freeswitch-dev] How to unbridge callers on detection of DTMF * key
 and send them back to where they were
Message-ID: <333789DE5C38474EB3A478A538F4EBAB098855F836@prod-exch01.corp.vseinc.com>

Hi,

I've been developing an application module in C/C++ for our online personals type of application.  All are well until I need to bridge two existing sessions together and need to find a way to monitor DTMF key (*) so I can un-bridge them and hopefully put them back to where they were.

Call flow is like:

Caller 1 sends recorded message to Caller 2 asking for live connection
Caller 1 in a loop of listening to wait music and checking Caller 2 response (via database)  (is there a better way like putting caller 1 on hold and somehow signal it from Caller 2's thread?)
Caller 2 hears message and accepts (update database)
*bridge Caller 1 and Caller 2
*monitor DTMF for asterisk key
*un-bridge and send them back to where they were so they can continue to exchange messages with other callers


I tried switch_ivr_uuid_bridge function like below:

Originator side:
switch_ivr_uuid_bridge

//set up args so it stops on any dtmf
while (switch_channel_ready(channel))
{
                status = switch_ivr_sleep(session, 60000, SWITCH_TRUE, &args);
                if (status == SWITCH_BREAK && dtmf_key == '*')
                                break;
}

Originatee side:
//set up args so it stops on any dtmf
while (switch_channel_ready(channel))
{
                status = switch_ivr_sleep(session, 60000, SWITCH_TRUE, &args);
                if (status == SWITCH_BREAK && dtmf_key == '*')
                                break;
}

The problem is after switch_ivr_uuid_bridge call, the channel state has changed and switch_channel_ready becomes false and I immediately got out of the loop and eventually exit from the thread.

How do I make this work?  Any suggestions are appreciated. Thanks.

Eddy Chu
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